Hello guys,
I'm trying to add an xml with a boundary to an outgoing INVITE. But if i do:
set_body_multipart("$rb", "application/sdp", "delimiter");
msg_apply_changes();
append_body_part("$var(something)", "application/pidf+xml");
msg_apply_changes();
but kamailio adds the SDP, but NOT the second $var(something). If i switch
them and add firs the var, and then the SDP. kamailio adds the contents of
$var(something) but NOT the SDP.... point is it doesn't add the second
append
Thanks!!!
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
hi,
i have kamailio gw (5.6.x) for voxbone. dispatcher module
voxbone has separated inbound/outboud and prohibit sip options to
"inbound" IP
#traffic from voxbone
20 sip:81.201.82.45:5060;transport=udp 1 0 duid=abc;socket=udp:x.x.x.x:5060
#traffic TO voxbone
30 sip:81.201.89.110:5060;transport=udp 0 0 duid=abc;socket=udp:x.x.x.x:5060
is it possible send sip options per GW with dispatcher module?
i tried flag "1 (bit at index 0 - 1 <<0) - inactive destination"
https://www.kamailio.org/docs/modules/devel/modules/dispatcher.html#idm1059
Marek
Hi all!
got a weird behavior that I cannot understand the reason for...
In our LAB environment, we have 2 Asterisk instances (version 13.38.3 and
chan_sip) and 1 Kamailio 5.7 in between.
All servers are in the same network, so, there is no NAT involved. No
RTPEngine neither.
Network is 10.20.0.0/24 and Asterisk #1 has IP .1 Asterisk #2 has IP .3
and Kamailio has IP .5
The Asterisk servers are used only for testing, nothing serious. However,
Kamailio is setup to use RTJson requesting routes to a Routing Server on
the same network. And it works fine.
Both Asterisk servers have the same dialplan, which only Answers the call
and plays MOH on both ends so that RTP audio streams both ways.
When making a call on Asterisk Server #1 via command line to go directly to
Asterisk Server #2 without using Kamailio (CLI> channel originate SIP/
123(a)10.20.0.*3* application MusicOnHold() ) the Asterisk #2 receives the
call, answers and plays MOH too and I can see RTP streams coming from both
ends correctly.
However, if I use Kamailio to proxy the call generated from Asterisk #1 to
Asterisk #2, using similar command line instruction (CLI> channel originate
SIP/123(a)10.20.0.5 application MusicOnHold() ), the call is indeed received
on Kamailio who then sends it to Asterisk #2, who answers the call and
plays MOH, *but* despite the audio stream being sent to Asterisk #1 it is
never received, however audio from Asterisk #1 is received by Asterisk #2,
which configures a typical One Way Audio issue due to NAT.
This is where it gets strange, because there is no NAT, SDP on INVITE and
SIP 200 messages seem OK, as far as I understand it.
Also, Asterisk servers have SIP configuration with directmedia enabled and
NAT disabled to make sure that media is direct. But I have also tried with
directmedia disabled and NAT enabled and get identical results.
I am most probably missing some tiny detail, but I have no clue.... and I
bet it is simple and stupid....
Could another pair of eyes help me with this? What is wrong? Do I really
need RTPEngine even when the network has no NAT? I am sure it would work
that way, but it doesn't make sense...
Here are some screenshots:
Call Scenario #1 - direct call from Asterisk #1 to Asterisk #2 without
Kamailio in between:
Invite from Asterisk #1 to Asterisk #2 with direct media between both ends:
[image: image.png]
Replies from Asterisk #2 to Asterisk #1 with direct media between both ends:
[image: image.png]
Call Scenario #2 - call from Asterisk #1 using Kamailio to relay call to
Asterisk #2, with one way audio
Invite from Asterisk #1 to Asterisk #2 via Kamailio with SDP details:
[image: image.png]
Invite from Kamailio relayed to Asterisk #2 with SDP details from Asterisk
#1 identical to above:
[image: image.png]
Reply from Asterisk #2 to Kamailio with SDP details:
[image: image.png]
Reply from Kamailio to Asterisk #2 with SDP details from Asterisk #2
identical to above:
[image: image.png]
As we can see, SDP details seem OK, but if I check call flow on Asterisk
#1, I can only find 1 RTP channel with audio coming from Asterisk #2
[image: image.png]
and the same on Asterisk #2 :
[image: image.png]
My Kamailio.cfg :
#!KAMAILIO
#
# config file for SIPProxy
# - load balancing of VoIP calls
# - no TPC listening
#
# Kamailio (OpenSER) SIP Server v3.2
# - web: http://www.kamailio.org
# - git: http://sip-router.org
#
#
# Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php
# 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
#
#!define WITH_DEBUG
###!define WITH_NAT
#!define WITH_PSTN
/* enables Accounting Log functions */
#!define FLT_ACC 1
/* enable Accounting of missed or failed calls */
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
/* defines DB connection string */
#!ifndef DBURL
#!define DBURL "mysql://kamailio:kamailio@10.20.0.1:3306/kamailio"
#!endif
# - the value for 'use_domain' parameters
#!define MULTIDOMAIN 1
####### Global Parameters #########
#!ifdef WITH_DEBUG
debug=4
log_stderror=no
#!else
debug=2
log_stderror=no
#!endif
#!define FLT_DISPATCH_SETID 1
#!define FLT_FS 10
#!define FLT_NATS 5
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7
#!define FLT_SRC_ALLOWED 8
#!define FLT_DST_INTERNAL_IP 9
#!define FLT_SRC_INTERNAL_IP 10
#!substdef "!INTERNAL_IP_NET!10.20.0.0/24!g"
#!substdef "!INTERNAL_IP_ADDR!10.20.0.2!g"
#!substdef "!EXTERNAL_IP_ADDR!10.20.0.2!g"
#!ifndef HTTP_ASYNC_CLIENT_WORKERS
#!define HTTP_ASYNC_CLIENT_WORKERS 8
#!endif
/* add API http timeout */
#!define HTTP_API_TIMEOUT 5000
#!define HTTP_API_ROUTING_ENDPOINT "http://10.246.212.40:7778/get_route"
/* DMQ SIP message sharing */
#!define DMQ_PORT 5062
#!define DMQ_LISTEN "sip:10.20.0.2:5062"
#!define DMQ_SERVER_ADDRESS "sip:10.20.0.2:5062"
#!define DMQ_NOTIFICATION_ADDRESS "sip:10.20.0.4:5062"
memdbg=5
memlog=5
log_facility=LOG_LOCAL0
log_prefix="{$mt $hdr(CSeq) $ci} "
fork=yes
children=8
/* comment the next line to enable TCP - all trunks are UDP only */
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
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:10.20.0.5:5060 advertise 10.20.0.5:5060
listen=tcp:10.20.0.5:5060 advertise 10.20.0.5:5060
listen=udp:10.20.0.2:5062
advertised_address="10.20.0.5";
sip_warning=no;
use_dns_failover = on;
####### Modules Section ########
#set module path
mpath="/usr/local/lib64/kamailio/modules/"
loadmodule "db_mysql.so"
loadmodule "jsonrpcs.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 "acc.so"
loadmodule "usrloc.so"
loadmodule "nathelper.so"
#loadmodule "rtimer.so"
#loadmodule "sqlops.so"
# --- CPS Limiter
# --- end of CPS Limiter
loadmodule "ipops.so"
loadmodule "textopsx.so"
loadmodule "sdpops.so"
loadmodule "http_async_client.so"
loadmodule "rtjson.so"
loadmodule "jansson.so"
loadmodule "dmq.so"
loadmodule "dmq_usrloc.so"
loadmodule "htable.so"
loadmodule "dialog.so"
#!ifdef WITH_DEBUG
loadmodule "debugger.so"
#!endif
#!ifdef WITH_DEBUG
# ----- debugger params -----
modparam("debugger", "log_level_name", "exec")
#!endif
# ----------------- setting module-specific parameters ---------------
# ----- jsonrpcs params -----
modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo")
modparam("jsonrpcs", "pretty_format", 1)
# ----- rr params -----
modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo")
modparam("jsonrpcs", "pretty_format", 1)
# ----- 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", "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=$si")
# ----- acc params -----
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc",
"log_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
/* enhanced DB accounting */
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", DBURL)
modparam("acc",
"db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
//;calltype=$avp(calltype)")
# ----- tm params -----
# ----- the TM module enables stateful processing of SIP requests
modparam("tm", "fr_timer", 5000)
modparam("tm", "fr_inv_timer", 60000)
modparam("tm", "remap_503_500", 0)
# ----- usrloc params -----
/* enable DB persistency for location entries */
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
# params needed for NAT traversal in other modules
modparam("usrloc", "nat_bflag", FLB_NATB)
# ----- nathelper params -----
modparam("nathelper", "received_avp", "$avp(s:rcv)")
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:ping@kamailio.org")
#modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;")
#modparam("rtimer", "exec", "timer=cdr;route=CDRS")
#modparam("sqlops", "sqlcon", "ca=>mysql://
kamailio:kamailiorw@10.19.139.113:3306/kamailio")
#modparam("dmq", "server_socket", DMQ_SERVER_SOCKET )
modparam("dmq", "server_address", DMQ_SERVER_ADDRESS )
modparam("dmq", "notification_address", DMQ_NOTIFICATION_ADDRESS )
modparam("dmq", "multi_notify", 1)
modparam("dmq", "num_workers", 4)
modparam("dmq", "ping_interval", 60)
modparam("dmq_usrloc", "enable", 1)
# -- CPS Limiter
modparam("htable", "htable", "rhs=>size=32;initval=0;autoexpire=10;")
modparam("htable", "htable", "rhm=>size=32;initval=0;autoexpire=120;")
modparam("htable", "enable_dmq", 1)
modparam("htable", "dmq_init_sync", 1)
modparam("dialog", "profiles_with_value", "concurrent_calls")
modparam("dialog", "enable_dmq", 1)
# ----- http_async_client params -----
modparam("http_async_client", "workers", HTTP_ASYNC_CLIENT_WORKERS)
modparam("http_async_client", "connection_timeout", 2000)
####### Routing Logic ########
# main request routing logic
route {
if (is_method("KDMQ") && $Rp == 5062)
{
dmq_handle_message();
}
xlog("L_INFO"," ********** Route START ***********");
# log the basic info regarding this call
xlog("L_INFO","start|\n");
xlog("L_INFO","===================================================\n");
xlog("L_INFO","New SIP message $rm with call-ID $ci \n");
xlog("L_INFO","---------------------------------------------------\n");
xlog("L_INFO"," received $pr request $rm $ou\n");
xlog("L_INFO"," source $si:$sp\n");
xlog("L_INFO"," from $fu\n");
xlog("L_INFO"," to $tu\n");
xlog("L_INFO","---------------------------------------------------\n");
xlog("L_INFO","---------------------------------------------------\n");
# OPTIONS requests without a username in the Request-URI but one
# of our domains or IPs are addressed to the proxy itself and
# can be answered statelessly.
if (is_method("OPTIONS"))
{
sl_send_reply("200","OK");
exit;
}
if ($fU=="ping")
{
sl_send_reply("200","OK");
exit;
}
# extract original source ip / port from X-forwarded-For header
route(HANDLE_X_FORWARDED_FOR);
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# handle requests within SIP dialogs
### only initial requests (no To tag)
# 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();
xlog("L_INFO", "ROUTE - Exiting after Retransmission check - method $rm");
exit;
}
t_check_trans();
}
route(WITHINDLG);
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
xlog("L_INFO", "ROUTE - Removing Headers");
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE")){
t_on_failure("MANAGE_FAILURE");
xlog("L_INFO", "ROUTE - Recording Route");
record_route();
if (is_method("INVITE") && is_request()) {
if (has_body("application/sdp")) {
xlog("L_INFO", "ROUTE - goiing to t_on_reply[ON_REPLY]\n");
t_on_reply("ON_REPLY");
}
}
}
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply("484","ROUTE - Address Incomplete");
exit;
}
route(TOCARRIER);
xlog("L_INFO", " ********** Route END *************");
}
# extract original source ip / port from X-forwarded-For header
route[HANDLE_X_FORWARDED_FOR] {
if (is_present_hf("X-Forwarded-For")) {
$var(source_ip) = $(hdr(X-Forwarded-For){s.select,0,:});
$var(source_port) = $(hdr(X-Forwarded-For){s.select,1,:});
} else {
$var(source_ip) = $si;
$var(source_port) = $sp;
}
$var(to_number) = $rU;
}
route[RELAY_API] {
xlog("L_INFO","RELAY_API - from_ip $var(source_ip):$var(source_port)
from_number $fU to_number $ru");
$http_req(all) = $null;
$http_req(suspend) = 1;
$http_req(timeout) = HTTP_API_TIMEOUT;
$http_req(method) = "POST";
$http_req(hdr) = "Content-Type: application/json";
jansson_set("string","from_ip",$var(source_ip), "$var(http_routing_query)");
jansson_set("string","from_port",$var(source_port),
"$var(http_routing_query)");
jansson_set("string","from_number",$fU, "$var(http_routing_query)");
jansson_set("string","to_number",$var(to_number) ,
"$var(http_routing_query)");
xlog("L_INFO","RELAY_API - API ASYNC ROUTING REQUEST:
$var(http_routing_query)\n");
$http_req(body) = $var(http_routing_query);
t_newtran();
http_async_query(HTTP_API_ROUTING_ENDPOINT, "RELAY_API_RESPONSE");
}
# Relay request using the API (response)
route[RELAY_API_RESPONSE] {
if ($http_ok==1 && $http_rs==200)
{
xlog("L_INFO","RELAY_API_RESPONSE - RESPONSE: $http_rb\n");
if (jansson_get("rtjson", $http_rb, "$var(rtjson)")) {
xlog("L_INFO","RELAY_API_RESPONSE - $var(rtjson)");
rtjson_init_routes("$var(rtjson)");
rtjson_push_routes();
# relay the message
t_on_branch("MANAGE_BRANCH");
t_on_failure("MANAGE_FAILURE");
route(RELAY);
return;
}
}
send_reply(500, "API Not Available - http response = $http_rs $http_ok");
exit;
}
onreply_route[ON_REPLY] {
xlog("L_INFO", "ON_REPLY - In onreply_route[ON_REPLY] $rs");
# on reply
if (t_check_status("183|180|200")) {
xlog("L_INFO", "ON_REPLY - Fixing Contacts");
# subst_hf("Contact","/@.*:/@EXTERNAL_IP_ADDR:/","a");
//subst_hf("Record-Route","/INTERNAL_IP_ADDR/EXTERNAL_IP_ADDR/","f");
}
if (has_body("application/sdp")){
if (sdp_remove_line_by_prefix("a=maxptime")){
xlog("L_INFO", "ON_REPLY - remove maxptime ");
msg_apply_changes();
}
else{
xlog("L_INFO", "ON_REPLY - did not removed maxptime ");
}
}
if (t_check_status("408")) {
xlog("L_INFO", "ROUTE - Handling 408 Timeout\n");
}
}
route[TOCARRIER]{
#using rtjson, unsomment following line
route(RELAY_API);
}
# Per SIP request initial checks
route[REQINIT] {
xlog("L_INFO", "REQINIT - Starting");
if (!mf_process_maxfwd_header("10")) {
xlog("L_INFO", "REQINIT - 483 - Too Many Hops");
sl_send_reply("483","Too Many Hops");
exit;
}
if(!sanity_check("1511", "7"))
{
xlog("L_INFO","REQINIT - Sanity Check -> Malformed SIP message from
$si:$sp\n");
exit;
}
}
# Caller NAT detection
route[NATDETECT] {
xlog("L_INFO", "NATDETECT - Entering");
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
xlog("L_INFO", "NATDETECT - Fix Nated Register");
fix_nated_register();
} else {
if(is_first_hop()){
xlog("L_INFO", "NATDETECT - Set Contact Alias");
set_contact_alias();
}
}
xlog("L_INFO", "NATDETECT - Set FLT_NATS" + FLT_NATS);
setflag(FLT_NATS);
}
#!endif
xlog("L_INFO", "NATDETECT - NAT Detect set FLT_NTS = " + FLT_NATS);
return;
}
# Handle requests within SIP dialogs
route[WITHINDLG] {
xlog("L_INFO", "WITHINDLG - Entering");
if (!has_totag()) return;
if (is_present_hf("Route") && $hdrc(Route)==1)
{
if (search_hf("Route", ".*EXTERNAL_IP_ADDR.*", "f"))
{
xlog("L_INFO", "WITHINDLG - Removing the route to self");
remove_hf("Route");
}
}
# sequential request within a dialog should
# take the path determined by record-routing
if (loose_route()) {
route(DLGURI);
if (is_method("BYE|CANCEL")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
}
else if ( is_method("ACK") ) {
# ACK is forwarded statelessy
xlog("L_INFO", "WITHINDLG - Going to NATMANAGE");
route(NATMANAGE);
}
else if ( is_method("NOTIFY") ) {
#Add Record-Route for in-dialog NOTIFY as per RFC 6665.
record_route();
}
if(is_method("BYE"))
xlog("L_INFO", "WITHINDLG - BYE message from $rU");
route(RELAY);
exit;
}
if ( is_method("ACK|BYE|INVITE|UPDATE") ) {
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. Try to route anyway - being optimistic
# since it has at least a To Tag
route(RELAY);
exit;
}
}
sl_send_reply("404","Not here");
xlog("L_INFO", "WITHINDLG - Finishing WITHINDLG");
exit;
}
# URI update for dialog requests
route[DLGURI] {
xlog("L_INFO", "WITHINDLG - Entering DLGURI");
#!ifdef WITH_NAT
if(!isdsturiset()) {
xlog("L_INFO", "WITHINDLG - Handle ruri ALIAS");
handle_ruri_alias();
}
#!endif
return;
}
# Routing to foreign domains ---> NOT USED
route[SIPOUT] {
xlog("L_INFO", "WITHINDLG - Entering SIPOUT");
if (uri==myself){
xlog("L_INFO", "WITHINDLG - URI is MySelf!");
return;
}
append_hf("P-hint: outbound\r\n");
xlog("L_INFO", "WITHINDLG - Finishing SIPOUT");
route(RELAY);
exit;
}
# Wrapper for relaying requests
route[RELAY] {
xlog("L_INFO", " ******** RELAY *******");
xlog("L_INFO", "RELAY - $si $su $ru");
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|CANCEL|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) {
xlog("L_INFO", "RELAY - branch_route NOT SET!");
t_on_branch("MANAGE_BRANCH");
}
}
xlog("L_INFO", "RELAY - checking method");
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
xlog("L_INFO", "RELAY - is INVITE|SUBSCRIBE|UPDATE");
if(!t_is_set("onreply_route")) {
xlog("L_INFO", "RELAY - onreply_route NOT SET!");
t_on_reply("ON_REPLY"); # MANAGE_REPLY");
}
}
if (is_method("INVITE")) {
xlog("L_INFO", "RELAY - is INVITE");
t_on_failure("FAILED_RELAY");
if(!t_is_set("failure_route")) {
xlog("L_INFO", "RELAY - failure_route NOT SET!");
t_on_failure("MANAGE_FAILURE");
}
}
if (!t_relay()) {
xlog("L_INFO", "RELAY - t_relay returns FALSE");
route("MANAGE_FAILURE");
#sl_reply_error();
}
xlog("L_INFO", "RELAY - exiting");
exit;
}
failure_route[FAILED_RELAY] {
xlog("L_INFO", "FAILED_RELAY - Entering");
if (t_check_status("[4-5][0-9][0-9]")){
xlog("L_INFO", "FAILED_RELAY - Could not reach destination endpoint!");
if (rtjson_next_route()) {
xlog("L_INFO", "MANAGE_FAILURE - Getting next route");
t_on_branch("MANAGE_BRANCH");
t_on_failure("MANAGE_FAILURE");
route(RELAY);
}
}
}
route[NATMANAGE] {
xlog("L_INFO", "NATMANAGE - Entering");
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
xlog("L_INFO", "NATMANAGE - nat=yes --- Setting FLB_NATB");
setbflag(FLB_NATB);
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) ))
{
xlog("L_INFO", "NATMANAGE - NO FLT_NATS/B Set!!! Getting out of
NATMANAGE");
return;
}
if (is_request()) {
xlog("L_INFO", "NATMANAGE - is_request - $rm from $si");
if (!has_totag()) {
if(t_is_branch_route()) {
xlog("L_INFO", "NATMANAGE - adding nat=yes");
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
xlog("L_INFO", "NATMANAGE - is_reply - $rm from $si");
if(isbflagset(FLB_NATB)) {
if(is_first_hop())
{
xlog("L_INFO", "NATMANAGE - Set Contact Alias");
set_contact_alias();
}
}
}
#!endif
return;
}
# Manage failure routing cases
route[MANAGE_FAILURE] {
xlog("L_INFO", "MANAGE_FAILURE - Entering ");
route(NATMANAGE);
xlog("L_INFO", "MANAGE_FAILURE - t_is_canceled");
if (t_is_canceled()) exit;
#!ifdef WITH_BLOCK3XX
# block call redirect based on 3xx replies.
if (t_check_status("3[0-9][0-9]")) {
xlog("L_INFO", "MANAGE_FAILURE - SIP 3XX returned!!");
t_reply("404","Not found");
exit;
}
#!endif
#!ifdef WITH_BLOCK401407
# block call redirect based on 401, 407 replies.
if (t_check_status("401|407")) {
xlog("L_INFO", "MANAGE_FAILURE - SIP 401|407 returned!!");
t_reply("404","Not found");
exit;
}
#!endif
if (t_check_status("503")){
xlog("L_INFO", "MANAGE_FAILURE - SIP 503 returned : no destination
available");
t_reply("503", "Destination not available");
exit;
}
if (rtjson_next_route()) {
xlog("L_INFO", "MANAGE_FAILURE - Getting next route!!");
t_on_branch("MANAGE_BRANCH");
t_on_failure("MANAGE_FAILURE");
route(RELAY);
exit;
}
}
# Manage outgoing branches
branch_route[MANAGE_BRANCH] {
xlog("L_INFO","MANAGE_BRANCH - New branch [$T_branch_idx] to
$ru\n");
xlog("L_INFO", "MANAGE_BRANCH - branch_route MANAGE_BRANCH 1 ");
rtjson_update_branch();
route(NATMANAGE);
}
Any help would be greatly appreciated.
Thanks in advance.
*Sérgio *
Dear all,
Is there any other possibility to (automatically) check the plausibility/syntax of an existing kamailio.cfg, besides
kamailio -c -f kamailio.cfg
?
Thanks,
Christoph
Hi all!
I'm testing the DMQ module, but having difficulties setting it up.
[...]
My kamailio.cfg is as follows :
[...]
listen=udp:0.0.0.0:5060
listen=tcp:0.0.0.0:5060
[...]
modparam("dmq", "server_address", "sip:10.20.0.2:5062") #DMQ_SERVER_ADDRESS
modparam("dmq", "notification_address", "sip:10.20.0.4:5062
")#DMQ_NOTIFICATION_ADDRESS
modparam("dmq", "multi_notify", 1)
modparam("dmq", "num_workers", 4)
modparam("dmq", "ping_interval", 15)
modparam("dmq_usrloc", "enable", 1)
If I check syntax (using -c parameter) all is ok. Once I start Kamailio,
the following error appears on system log:
INFO: dmq [dmq.c:164]: make_socket_str_from_uri(): unknown transport
protocol - fall back to udp
ERROR: dmq [dmq.c:241]: mod_init(): server_uri is not a socket the proxy is
listening on
ERROR: <core> [core/sr_module.c:1030]: init_mod(): Error while initializing
module dmq (/usr/local/lib64/kamailio/modules/dmq.so)
I have used UDP as well as SIP as protocols for Server Address and
Notification Address, used port 5060 and 5062 (and others too), even added
a new listen=udp:0.0.0.0:5062, as well as listen=tcp:0.0.0.0:5062 but the
result is always the same.
Using Kamailio 5.7.4 on RHEL 9 with SELINUX set to enforcing (if relevant).
No firewall active.
local IP address is 10.20.0.1
second kamailio node IP address is 10.20.0.4
Any ideas? At first sight, it seems to be all set as per documentation.
I even found other configurations online, and apart from the IP address,
all is similar.
Thanks in advance,
*Sérgio Charrua*
*www.voip.pt <http://www.voip.pt/>*
