Hi all!
some additional details for this issue.
Currently, Kamailio is using RTJSON to get routes from the routing engine
and forward calls to the correct route.
Please note that the 2 testing endpoints and Kamailio are all in the same
network, no NAT involved, and firewalls are disabled!
Following route function does the magic:
route[TOCARRIER]{ #Route to send calls to a carrier at 192.168.200.130
route(RELAY_API); #Route relay
}
route[RELAY_API]{
# makes the HTTP Assync request
.....
# once response is received from HTTP REST API, go to 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;
}
}
}
This is working correctly.
However, as mentioned in previous email, when the call is forwarded to the
endpoint using RTJSON module (and for testing purposes, we are using
Asterisk 13.38.x as an endpoint), it results in a one-way audio issue: A
Leg sends Audio Streams correctly directly to B Leg (direct media) but B
Leg seems to not sending any audio, even though both endpoints are playing
some Music On Hold stuff.
Even TCPDUMP shows no RTP traffic from B to A, but can find traffic from A
to B!
What I found out is that if I modify RELAY_API route to be as follows:
route[TOCARRIER]{
rewritehost("10.20.0.3"); #Rewrite host to be the endpoint's IP
route(RELAY);
}
The audio streams are fully working, both ways! TCPDUMP shows audio traffic
both ways, no issues!
The SIP Traces show the same structure, both for SIP and SDP (of course,
CallID, BranchID and RURI are different), so I think the issue is *not*
within endpoints, but somewhere in Kamailio (module or configuration).
Any suggestions?
*Sérgio Charrua*
On Tue, Mar 5, 2024 at 1:05 PM Sergio Charrua <sergio.char...@voip.pt>
wrote:
> 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 either.
> 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@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@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:
>
> https://drive.google.com/file/d/1eLjT3nr_Rc-UBaf4QhIgZ95bjETOVvxo/view?usp=drive_link
>
> Replies from Asterisk #2 to Asterisk #1 with direct media between both
> ends:
>
> https://drive.google.com/file/d/11lLcB-V8rWGSrVqWiit-q9WX2FfqB6BZ/view?usp=drive_link
>
> 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:
>
> https://drive.google.com/file/d/1Cp9xrGcwNmQ9Ks36N_oD1Dj7lxfu-tbH/view?usp=drive_link
>
> Invite from Kamailio relayed to Asterisk #2 with SDP details from
> Asterisk #1 identical to above:
>
> https://drive.google.com/file/d/1mi3FCNjM3luXfENEp-0088XLgcyfXRK6/view?usp=drive_link
>
> Reply from Asterisk #2 to Kamailio with SDP details:
>
> https://drive.google.com/file/d/1TpMGe2tvpX_5SIpSbm2b3Zro9EuYwcO-/view?usp=drive_link
>
> Reply from Kamailio to Asterisk #2 with SDP details from Asterisk #2
> identical to above:
>
> https://drive.google.com/file/d/12jq5APfFwVVPc0vJ3RkcXyN1hBE51fnQ/view?usp=drive_link
>
> 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
>
> https://drive.google.com/file/d/1iEfvkylZVbthHM5kxkurWh-GAYCYLytl/view?usp=drive_link
>
> and the same on Asterisk #2 :
>
> https://drive.google.com/file/d/12rvf9Lrwp-MNZvGCwlXEEBsBRmfcinph/view?usp=drive_link
>
> My Kamailio.cfg is as follows:
>
> #!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:p...@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 Charrua
>
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