[SR-Users] Problem trying to play recorded message before call

Carsten Bock carsten at ng-voice.com
Fri Nov 18 09:56:59 CET 2011


Hi,

the RTPProxy does not play wav-files directly. You need to convert
them into a custom rtpproxy-format using "makeann", which is part of
the rtpproxy-package.
Makeann encodes the file directly into PCMA/PCMU/GSM-Codec files.

Also: To whom do you want to play the announcement? To the Calller?
The rtpproxy does not create a proper response towards your phone
(e.g. a "183 Session progress"), it just inserts the announcement into
an existing stream (e.g. when the call is established, for music on
hold).

Carsten


2011/11/17 David Candamil Santos <candamil at gmail.com>:
> Hi there, guys. I hope you can help me. I use kamailio to comunicate some
> servers (each one with its own kamailio). When an user calls another,
> depending on the userID that will receive the call, the server redirects the
> call to the corresponding kamailio server. It works properly. Now, what I
> would like to do is to configure it so it plays a notification saying where
> the call is going to be forwarded.
> After reading some tutorials, I tried to get it working with rtpproxy, but
> after configuring it, it doesn't play the clips. I am testing it with a
> *.wav file. The path to the file is "/usr/local/etc/kamailio/test.wav" (it's
> in the kamailio directory). This is my software:
> Debian stable
> kernel 2.6.32-5-686
> Kamailio 3.2.0 (GIT installation)
> rtpproxy 1.2.1-1
> Kamailio and rtpproxy are running in the same machine.
> These are the log messages:
> -------------------kamailio.log-------------------
> (SEVERAL TIMES)
> Nov 16 12:58:10 debian-virtualbox kamailio[2076]: INFO: rtpproxy
> [rtpproxy.c:1415]: rtp proxy <udp:127.0.0.1:22222> found, support for it
> enabled
> (SEVERAL TIMES)
> Nov 16 12:58:12 debian-virtualbox kamailio[2069]: ERROR: <script>:
> Intentando reproducir audio
> --------------------------------------------------
>
> This is the relevant configuration:
> --------------/etc/default/rtpproxy---------------
> # The control socket.
> #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock"
> # To listen on an UDP socket, uncomment this line:
> CONTROL_SOCK=udp:127.0.0.1:22222
> # Additional options that are passed to the daemon.
> EXTRA_OPTS="-l 127.0.0.1"
> --------------------------------------------------
>
> -----/usr/local/etc/kamailio/kamailio.cfg (just my important changes for
> this action)---------
> loadmodule "rtpproxy.so"
> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222")
> request_route {
>
> rtpproxy_offer();
> xlog("L_ERR","Intentando reproducir audio");
> rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav","1");
> (...default routes ommited...)
> route(RELAY);
> }
> --------------------------------------------------
> I can play it with mplayer, so it's not a sound problem (or even a system
> codec problem). I would like to know if I can make it working with rtpproxy,
> or if there is an easier way to do it. Thanks for your time.
> (I include now the whole kamailio.cfg, just in case you need to check
> anything)
> ---------/usr/local/etc/kamailio/kamailio.cfg (complete) ----------
> #!KAMAILIO
> #
> # Kamailio (OpenSER) SIP Server v3.2 - 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
> #
> ##--
> # WITH_DEBUG
> # *** To enable mysql:
> #     - define WITH_MYSQL
> #
> ##--
> #!define WITH_MYSQL
> # *** To enable authentication execute:
> #     - enable mysql
> #     - define WITH_AUTH
> #     - add users using 'kamctl'
> #
> ##--
> #!define WITH_AUTH
> # *** 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:
> #     - enable 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
> #
> ##--
> #WITH_NAT
> # *** 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 VARCHAR(128) NOT NULL
> DEFAULT '';
> #!endif
> ####### Defined Values #########
> # *** 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://openser:openserrw@localhost/openser"
> #!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_ACCFAILED 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.mydomain.com"
> /* 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:10.0.0.10:5060
> /* port to listen to
>  * - can be specified more than once if needed to listen on many ports */
> port=5060
> #$banana_server="192.168.0.70"
> #$treasure_server="192.168.0.60"
> #$drake_server="192.168.0.60"
> #!ifdef WITH_TLS
> enable_tls=yes
> #!endif
> # 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
> ####### Modules Section ########
> # set paths to location of modules (to sources or installation folders)
> #!ifdef WITH_SRCPATH
> mpath="modules_k:modules"
> #!else
> mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/"
> #!endif
> #!ifdef WITH_MYSQL
> loadmodule "db_mysql.so"
> #!endif
> ##--
> loadmodule "sdpops.so"
> loadmodule "textopsx.so"
> loadmodule "rtpproxy.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 "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"
> #!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
> # ----------------- setting module-specific parameters ---------------
> ##--
> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222")
> # ----- 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)
> modparam("rr", "append_fromtag", 0)
>
> # ----- 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)
>
> # ----- 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=$rd")
> #!endif
>
> # ----- usrloc params -----
> /* enable DB persistency for location entries */
> #!ifdef WITH_USRLOCDB
> modparam("usrloc", "db_url", DBURL)
> modparam("usrloc", "db_mode", 2)
> modparam("usrloc", "use_domain", MULTIDOMAIN)
> #!endif
>
> # ----- auth_db params -----
> #!ifdef WITH_AUTH
> modparam("auth_db", "db_url", DBURL)
> modparam("auth_db", "calculate_ha1", yes)
> modparam("auth_db", "password_column", "password")
> modparam("auth_db", "load_credentials", "")
> modparam("auth_db", "use_domain", MULTIDOMAIN)
> # ----- permissions params -----
> #!ifdef WITH_IPAUTH
> modparam("permissions", "db_url", DBURL)
> modparam("permissions", "db_mode", 1)
> #!endif
> #!endif
>
> # ----- alias_db params -----
> #!ifdef WITH_ALIASDB
> modparam("alias_db", "db_url", DBURL)
> modparam("alias_db", "use_domain", MULTIDOMAIN)
> #!endif
>
> # ----- speedial params -----
> #!ifdef WITH_SPEEDDIAL
> modparam("speeddial", "db_url", DBURL)
> modparam("speeddial", "use_domain", MULTIDOMAIN)
> #!endif
>
> # ----- domain params -----
> #!ifdef WITH_MULTIDOMAIN
> modparam("domain", "db_url", DBURL)
> # use caching
> modparam("domain", "db_mode", 1)
> # register callback to match myself condition with domains list
> modparam("domain", "register_myself", 1)
> #!endif
>
> #!ifdef WITH_PRESENCE
> # ----- presence params -----
> modparam("presence", "db_url", DBURL)
> # ----- presence_xml params -----
> modparam("presence_xml", "db_url", DBURL)
> modparam("presence_xml", "force_active", 1)
> #!endif
>
> #!ifdef WITH_NAT
> # ----- rtpproxy params -----
> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222")
> # ----- nathelper params -----
> modparam("nathelper", "natping_interval", 30)
> modparam("nathelper", "ping_nated_only", 1)
> modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
> modparam("nathelper", "sipping_from", "sip:pinger at kamailio.org")
> # params needed for NAT traversal in other modules
> modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
> modparam("usrloc", "nat_bflag", FLB_NATB)
> #!endif
>
> #!ifdef WITH_TLS
> # ----- tls params -----
> modparam("tls", "config", "/usr/local/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 htable 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 {
> #start_recording();
> #force_rtp_proxy();
>      #rtpproxy_offer();
> #rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav", "1");
> #("fichero","numero de veces");
> #formato alaw?
>
> #force_rtp_proxy();
> rtpproxy_offer();
> xlog("L_ERR","Intentando reproducir audio");
> rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav","1");
>
>
>
> if ($rU=~"^001788[0-3].*") {
>        xlog("L_ERR", "Banana island");
> $rd="192.168.0.70";
> msg_apply_changes();
> xlog("L_ERR","Redireccionando a $rd\n");
>         }
> if ($rU=~"^001788[4-5].*") {
>        xlog("L_ERR", "Big treasure island");
> $rd="192.168.0.60";
> msg_apply_changes();
> xlog("L_ERR","Redireccionando a $rd\n");
>         }
> if ($rU=~"^001788[6-7].*") {
>        xlog("L_ERR", "Drake island");
> $rd="192.168.0.60";
> msg_apply_changes();
> xlog("L_ERR","Redireccionando a $rd\n");
>         }
> if ($rU=~"^001788[8-9].*") {
>        xlog("L_ERR", "Lost island");
> sdp_keep_codecs_by_name("GSM");
> #if para si true, bien, y si false mensaje y sale
>         }
>
> # per request initial checks
> route(REQINIT);
> # NAT detection
> route(NATDETECT);
> # 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();
> # 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);
> ##--
> #rtpproxy_offer();
>         #rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav", "1");
>
> route(RELAY);
> }
>
> route[RELAY] {
> # 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;
> }
> # 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();
> 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
> $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] {
> #!ifdef WITH_AUTH
> if (is_method("REGISTER"))
> {
> # authenticate the REGISTER requests (uncomment to enable auth)
> if (!www_authorize("$td", "subscriber"))
> {
> www_challenge("$td", "0");
> exit;
> }
> if ($au!=$tU)
> {
> sl_send_reply("403","Forbidden auth ID");
> exit;
> }
> } else {
> #!ifdef WITH_IPAUTH
> if(allow_source_address())
> {
> # source IP allowed
> return;
> }
> #!endif
> # authenticate if from local subscriber
> if (from_uri==myself)
> {
> if (!proxy_authorize("$fd", "subscriber")) {
> proxy_challenge("$fd", "0");
> exit;
> }
> if (is_method("PUBLISH"))
> {
> if ($au!=$fU || $au!=$tU) {
> sl_send_reply("403","Forbidden auth ID");
> exit;
> }
> if ($au!=$rU) {
> sl_send_reply("403","Forbidden R-URI");
> exit;
> }
> #!ifdef WITH_MULTIDOMAIN
> if ($fd!=$rd) {
> sl_send_reply("403","Forbidden R-URI domain");
> exit;
> }
> #!endif
> } else {
> if ($au!=$fU) {
> sl_send_reply("403","Forbidden auth ID");
> exit;
> }
> }
> consume_credentials();
> # caller authenticated
> } else {
> # caller is not local subscriber, then check if it calls
> # a local destination, otherwise deny, not an open relay here
> if (!uri==myself)
> {
> sl_send_reply("403","Not relaying");
> exit;
> }
> }
> }
> #!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
> }
> --------------------------------------------------------------
>
> _______________________________________________
> 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
>
>



-- 
Carsten Bock
CEO (Geschäftsführer)

ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany

http://www.ng-voice.com
mailto:carsten at ng-voice.com

Mobile +49 179 2021244
Office +49 40 34927219
Fax +49 40 34927220

Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock
Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/



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