[SR-Users] No audio when connect from public network but it works for lan users
CK Lee
ckleea at gmail.com
Sun Jan 4 13:39:25 CET 2015
I have this already in my sip.conf
[kamailio-ns]
type=friend
host=sip.example.org
port=5060
disallow=all
allow=gsm
allow=g729
allow=alaw
allow=ulaw
context=default
canreinvite=no
insecure=port,invite
dtmfmode=rfc2833
qualify=yes
nat=yes
I use asterisk 11 and the setup is ok without any one way audio issue.
On Sun, Jan 4, 2015 at 8:22 PM, Amit Patkar <amit at avhan.com> wrote:
> Please check sip.conf. You need to enable NAT options. Asterisk need to
> publish public IP in SDP for RTP traffic to reach your network. Asterisk
> need to differentiate local clients and external clients. localnet and
> externip parameters should be configured correctly.
>
> Regards,
> Amit Patkar
>
> On January 4, 2015 5:53:53 AM IST, CK Lee <ckleea at gmail.com> wrote:
>>
>> I am new to Kamailio after starting to play around for 3 weeks.
>>
>> Before Kamailio, I used asterisk and it works quite well in my network
>> which has 2 public IP addresses and 10+ wired and wireless users. The
>> asterisk listens to only one IP which my netgear router port forward the
>> necessary sip and rdp ports to the asterisk.
>>
>> My interest is to use Kamailio as SIP proxy to sit in front of the
>> asterisk. I follow the tutorial here -
>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
>>
>> I use debian wheezy and kamailio is installed from the repository with
>> version 4.2.1
>>
>> Everything works for my lan subscribers. However, when I am outside the
>> lan, I can register but do not the incoming packets at my sip client.
>>
>> What I observe is two str ange behaviours
>> 1. in asterisk, the kamailio peer has to use "sip.example.org or its
>> public IP" in the host name field, otherwise it is not online
>> 2. in looking at the log, I found the sip client peer is registered at my
>> lan address i.e. 192.168.118.xx instead of a public IP when I directly
>> login to asterisk.
>>
>> I add on rtproxy but not succeed with error as force_rtp_proxy_body:
>> incorrect port 0 in reply from rtpproxy
>>
>> Below is my kamailio.cfg
>>
>> Help is appreciated.
>> --------------------------------------
>> #!KAMAILIO
>>
>>
>>
>> #
>> # Kamailio (OpenSER) SIP Server v4.0 - default configuration script
>> # - web: http://www.kamailio.org
>> # - git: http://sip-router.org
>> #
>> # Direct your questions about this file to: <
>> sr-users at lists.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
>> #
>> # *** To enable mysql:
>> # - define WITH_MYSQL
>> #
>> # *** To enable authentication execute:
>> # - enable mysql
>> # - define WITH_AUTH
>> # - add users using 'kamctl'
>> #
>> # *** To enable IP authentication execute:
>> # - enable mysql
>> # - enable authentication
>> # - define WITH_IPAUTH
>> # - add IP addresses with group id '1' to 'address' table
>> #
>> # *** To enable persistent user location execute:
>> # - enabl e mysql
>>
>> # - define WITH_USRLOCDB
>> #
>> # *** To enable presence server execute:
>> # - enable mysql
>> # - define WITH_PRESENCE
>> #
>> # *** To enable nat traversal execute:
>> # - define WITH_NAT
>> # - install RTPProxy: http://www.rtpproxy.org
>> # - start RTPProxy:
>> # rtpproxy -l _your_public_ip_ -s udp:localhost:7722
>> #
>> # *** To enable PSTN gateway routing execute:
>> # - define WITH_PSTN
>> # - set the value of pstn.gw_ip
>> # - check route[PSTN] for regexp routing condition
>> #
>> # *** To enable database aliases lookup execute:
>> # - enable mysql
>> # - define WITH_ALIASDB
>> #
>> # *** To enable speed dial lookup execute:
>> # - enable mysql
>> # - define WITH_SPEEDDIAL
>> #
>> # *** To enable multi-domain support execute:
>> # - enable mysql
>> # - define WITH_MULTIDOMAIN
>> #
>> # *** To enable TLS support execute:
>> # - adjust CFGDIR/tls.cfg as needed
>> # - define WITH_TLS
>> #
>> # *** To enable XMLRPC support execute:
>> # - define WITH_XMLRPC
>> # - adjust route[XMLRPC] for access policy
>> #
>> # *** To enable anti-flood detection execute:
>> # - adjust pike and htable=>ipban settings as needed (default is
>> # block if more than 16 requests in 2 seconds and ban for 300
>> seconds)
>> # - define WITH_ANTIFLOOD
>> #
>> # *** To block 3XX redirect replies execute:
>> # - define WITH_BLOCK3XX
>> #
>> # *** To enable VoiceMail routing execute:
>> # - define WITH_VOICEMAIL
>> # - set the value of voicemail.srv_ip
>> # - adjust the value of voicemail.srv_port
>> #
>> # *** To enhance accounting execute:
>> # - enable mysql
>> # - define WITH_ACCDB
>> # - add following columns to database
>> #!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 VARCH AR(128) NOT NULL
>> DEFAULT '';
>>
>> #!endif
>>
>> ####### Defined Values #########
>>
>> #!define WITH_MYSQL
>> #!define WITH_AUTH
>> #!define WITH_USRLOCDB
>> #!define WITH_ASTERISK
>> #!define WITH_NAT
>> #!define WITH_TLS
>>
>> #!define ADDR_IPV4 192.168.118.30
>>
>> # *** Value defines - IDs used later in config
>> #!ifdef WITH_MYSQL
>> # - database URL - used to connect to database server by modules such
>> # as: auth_db, acc, usrloc, a.s.o.
>> #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio"
>> #!ifdef WITH_ASTERISK
>> #!define DBASTURL "mysql://user: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_ACCFA ILED 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
>>
>> fork=yes
>> children=4
>>
>> /* uncomment the next line to disable TCP (default on) */
>> #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
>>
>> /* add local domain aliases */
>> alias="sip.example.org"
>>
>> /* 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:192.168.118.30:5060
>> listen=tls:192.168.118.30:5061
>>
>> /* port to listen to
>> * - can be specified more than once if needed to listen on many ports */
>>
>> port=5060
>>
>> #!ifdef WITH_TLS
>> enable_tls=yes
>> #!endif
>>
>> #!ifdef WITH_XCAPSRV
>> tcp_accept_no_cl=yes
>> #!endif
>>
>>
>>
>> tcp_connection_lifetime=3604
>> tcp_accept_no_cl=yes
>> tcp_rd_buf_size=16384
>>
>> # 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 #########
>>
>> # These parameters can be modified runtime via RPC interface
>> # - see the documentation of 'cfg_rpc' module.
>> #
>> # Format: group.id = value 'desc' description
>> # Access: $sel(cfg_get.group.id) or @cfg_get.group.id
>> #
>>
>> #!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
>> asterisk.bindip = "192.168.118.30" desc "Asterisk IP Address"
>> asterisk.bindport = "15066" desc "Asterisk Port"
>> kamailio.bindip = "192.168.118.30" 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_k:modules"
>> #!else
>> mpath="/usr/lib/i386-linux-gnu/kamailio/modules/"
>> #!endif
>>
>> #!ifdef WITH_MYSQL
>> loadmodule "db_mysql.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"
>> loadmodule "outbound.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
>>
>> # ----------------- setting module-specific parameters ---------------
>>
>> modparam("ctl", "user", "kamailio")
>>
>> # ----- 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)
>> /* uncomment the next line not to allow more than 10 contacts per AOR */
>> #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:127.0.0.1:22222")
>> #modparam("rtpproxy", "rtpproxy_sock",
>> "unix:/var/run/rtpproxy/rtpproxy.sock")
>>
>> # ----- 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", "/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 h table 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
>>
>> ####### 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)
>>
>> Â � # 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] {
>>
>> #!ifdef WITH_NAT
>> if (check_route_param("nat=yes")) {
>> setbflag(FLB_NATB);
>> }
>> if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) {
>> route(RTPPROXY);
>> }
>> #!endif
>>
>> # 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;
>> }
>>
>>
>> # RTPProxy control
>> route[RTPPROXY] {
>> #!ifdef WITH_NAT
>> if (is_method("BYE")) {
>> unforce_rtp_proxy();
>> } else if (is_method("INVITE")){
>> rtpproxy_offer();
>> }
>> if (!has_totag()) add_rr_param(";nat=yes");
>> #!endif
>> return;
>> }
>>
>>
>> # 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 - se nd 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] {
>>
>> # 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;
>> }
>>
>> #!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("$fd", "sipusers", "1")) {
>> #!else
>> if (!auth_check("$fd", "subscriber", "1")) {
>> #!endif
>> auth_challenge("$fd", "0");
>> exit;
>> }
>> # user authenticated - remove auth header
>> if(!is_method("REGISTER|PUBLISH"))
>> consume_credentials();
>> }
>> #!endif
>> return;
>> }
>>
>> # Caller NAT detection route
>> route[NATDETECT] {
>> #!ifdef WITH_NAT
>> force_rport();
>> if (nat_uac_test("19")) {
>> if (is_method("REGISTER")) {
>> fix_nated_register();
>> } else {
>> fix_nated_contact();
>> }
>> setflag(FLT_NATS);
>> }
>> #!endif
>> return;
>> }
>>
>> # RTPProxy control
>> route[NATMANAGE] {
>> #!ifdef WITH_NAT
>> if (is_request()) {
>> if(has_totag()) {
>> if(check_route_param("nat=yes")) {
>> setbflag(FLB_NATB);
>> }
>> }
>> }
>> if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
>> return;
>>
>> rtpproxy_manage();
>>
>> if (is_request()) {
>> if (!has_totag()) {
>> add_rr_param(";nat=yes");
>> }
>> }
>> if (is_reply()) {
>> if(isbflagset(FLB_NATB)) {
>> fix_nated_contact();
>> }
>> }
>> #!endif
>> return;
>> }
>>
>> # Routing to foreign domains
>> route[SIPOUT] {
>> if (!uri==myself)
>> {
>> append_hf("P-hint: outbound\r\n");
>> route(RELAY);
>> }
>> }
>>
>> # PSTN GW routing
>> route[PSTN] {
>> #!ifdef WITH_PSTN
>> # check if PSTN GW IP is defined
>> if (strempty($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 $ru\n");
>> route(NATMANAGE);
>> }
>>
>> # manage incoming replies
>> onreply_route[MANAGE_REPLY] {
>> xdbg("incoming reply\n");
>> if(status=~"[12][0-9][0-9]")
>> route(NATMANAGE);
>> }
>>
>> # manage failure routing cases
>> failure_route[MANAGE_FAILURE] {
>> route(NATMANAGE);
>>
>> if (t_is_canceled()) {
>> exit;
>> }
>>
>> #!ifdef WITH_BLOCK3XX
>> # block call redirect based on 3xx replies.
>> if (t_check_status("3[0-9][0-9]")) {
>> t_reply("404","Not found");
>> exit;
>> }
>> #!endif
>>
>> #!ifdef WITH_VOICEMAIL
>> # serial forking
>> # - route to voicemail on busy or no answer (timeout)
>> if (t_check_status("486|408")) {
>> route(TOVOICEMAIL);
>> exit;
>> }
>> #!endif
>> }
>>
>> #!ifdef WITH_ASTERISK
>> # Test if coming from Asterisk
>> route[FROMASTERISK] {
>> if($si==$sel(cfg_get.asterisk.bindip)
>> && $sp==$sel(cfg_get.asterisk.bindport))
>> return 1;
>> return -1;
>> }
>>
>> # Send to Asterisk
>> route[TOASTERISK] {
>> $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":"
>> + $sel(cfg_get.asterisk.bindport);
>> route(RELAY);
>> exit;
>> }
>>
>> # Forward REGISTER to Asterisk
>> route[REGFWD] {
>> if(!is_method("REGISTER"))
>> {
>> return;
>> }
>> $var(rip) = $sel(cfg_get.asterisk.bindip);
>> $uac_req(method)="REGISTER";
>> $uac_req(ruri)="sip:" + $var(rip) + ":" +
>> $sel(cfg_get.asterisk.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();
>> }
>> #!endif
>>
>> ------------------------------
>>
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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