[Serusers] PSTN Configuration
Alberto Cruz
acruz at tekbrain.com
Tue Jun 28 22:31:28 CEST 2005
You can take a look at www.onsip.org. There is a detailed configuration
for PSTN integration
Regards
Alberto Cruz
Abdul Lateef wrote:
>Hello,
>
>I read lot of example ducumentations from net about
>the ser configuration. I found one example from
>voip-info.org. i configured the ser.cfg file
>accordidng to the guide. but when i am trying to
>restart the ser using /usr/local/sbin/serctl restart i
>am getting some pid file error messate like :
>
>----------------------------------
>[root at localhost root]# /usr/local/sbin/serctl stop
>Stopping SER : No PID file found! SER probably not
>running
>[root at localhost root]# /usr/local/sbin/serctl start
>
>Starting SER : PID file /var/run/ser.pid does not
>exist -- SER start failed
>[root at localhost root]#
>
>--------------------------------
>
>Here is my ser.cfg file configurations.
>I wanted to explain how i am going to use it.
>I have one VoIP Gatekeeper (MVTS) Which supports to
>register SER i want to register the SER in MVTS and
>the IP Phones i want to register in SER. So the call
>can be pass to destination using like:
>
>IP Phone > SER > MVTS > Destination PSTN.
>
>MVTS IP : 195.22.146.20
>SER : 10.0.0.20
>IP Phone : Should use the ser IP and uid/pws
>--------------- ser.cfg ----------------------
>
>#
> # $Id: pstn.cfg,v 1.2 2003/06/03 03:18:12 jiri Exp $
> #
> #
>
> # ------------------ module loading
>----------------------------------
>
> loadmodule "modules/sl/sl.so"
> loadmodule "modules/tm/tm.so"
> loadmodule "modules/acc/acc.so"
> loadmodule "modules/rr/rr.so"
> loadmodule "modules/maxfwd/maxfwd.so"
> loadmodule "modules/mysql/mysql.so"
> loadmodule "modules/auth/auth.so"
> loadmodule "modules/auth_db/auth_db.so"
> loadmodule "modules/group/group.so"
> loadmodule "modules/uri/uri.so"
>
> # ----------------- setting module-specific
>parameters ---------------
>
> modparam("auth_db",
>"db_url","sql://ser:heslo@10.0.0.26/ser")
> modparam("auth_db", "calculate_ha1", yes)
> modparam("auth_db", "password_column", "password")
>
> # — acc params --
> modparam("acc", "log_level", 1)
> # that is the flag for which we will account — don't
>forget to
> # set the same one :-)
> modparam("acc", "log_flag", 1 )
>
> # ------------------------- request routing logic
>-------------------
>
> # main routing logic
>
> route{
>
> /* ********* ROUTINE CHECKS
>********************************** */
>
> # filter too old messages
> if (!mf_process_maxfwd_header("10")) {
> log("LOG: Too many hops\n");
> sl_send_reply("483","Too Many Hops");
> break;
> };
> if (len_gt( max_len )) {
> sl_send_reply("513", "Wow — Message too
>large");
> break;
> };
>
> /* ********* RR
>********************************** */
>
> /* grant Route routing if route headers present
>*/
> if (loose_route()) { t_relay(); break; };
>
> /* record-route INVITEs — all subsequent
>requests must visit us */
> if (method=="INVITE") {
> record_route();
> };
>
> # now check if it really is a PSTN destination
>which should be handled
> # by our gateway; if not, and the request is an
>invitation, drop it --
> # we cannot terminate it in PSTN; relay
>non-INVITE requests — it may
> # be for example BYEs sent by gateway to call
>originator
> if (!uri=~"sip:\+?[0-9]+ at .*") {
> if (method=="INVITE") {
> sl_send_reply("403", "Call
>cannot be served here");
> } else {
> forward(uri:host, uri:port);
> };
> break;
> };
>
> # account completed transactions via syslog
> setflag(1);
>
> # free call destinations ... no authentication
>needed
> if ( is_user_in("Request-URI", "free-pstn") /*
>free destinations */
> |
>uri=~"sip:[79][0-9][0-9][0-9]@.*" /* local PBX */
> |
>uri=~"sip:98[0-9][0-9][0-9][0-9]") {
> log("free call");
> } else if (src_ip==192.168.0.10) {
> # our gateway doesn't support digest
>authentication;
> # verify that a request is coming from
>it by source
> # address
> log("gateway-originated request");
> } else {
> # in all other cases, we need to check
>the request against
> # access control lists; first of all,
>verify request
> # originator's identity
>
> if (!proxy_authorize( "gateway" /*
>realm */,
> "subscriber" /* table
>name */)) {
> proxy_challenge( "gateway" /*
>realm */, "0" /* no qop */ );
> break;
> };
>
> # authorize only for INVITEs —
>RR/Contact may result in weird
> # things showing up in d-uri that would
>break our logic; our
> # major concern is INVITE which causes
>PSTN costs
>
> if (method=="INVITE") {
>
> # does the authenticated user
>have a permission for local
> # calls (destinations beginning
>with a single zero)?
> # (i.e., is he in the "local"
>group?)
> if (uri=~"sip:0[1-9][0-9]+ at .*")
>{
> if
>(!is_user_in("credentials", "local")) {
>
>sl_send_reply("403", "No permission for local calls");
> break;
> };
> # the same for long-distance
>(destinations begin with two zeros")
> } else if
>(uri=~"sip:00[1-9][0-9]+ at .*") {
> if
>(!is_user_in("credentials", "ld")) {
>
>sl_send_reply("403", " no permission for LD ");
> break;
> };
> # the same for international
>calls (three zeros)
> } else if
>(uri=~"sip:000[1-9][0-9]+ at .*") {
> if
>(!is_user_in("credentials", "int")) {
>
>sl_send_reply("403", "International permissions
>needed");
> break;
> };
> # everything else (e.g., interplanetary calls) is
>denied
> } else {
> sl_send_reply("403",
>"Forbidden");
> break;
> };
>
> }; # INVITE to authorized PSTN
>
> }; # authorized PSTN
>
> # if you have passed through all the checks,
>let your call go to GW!
>
> rewritehostport("192.168.0.10:5060");
>
> # forward the request now
> if (!t_relay()) {
> sl_reply_error();
> break;
> };
>
> }
>
>
>
>
>
>Yours,
>Abdul Lateef
>Computer Programmer
>HATIF COM
>Mob: +974 - 5405022
>Tel: +974 - 4883068
>ICQ: 276994704
>YM!: abdul_zu
>Fax: +974 - 4883063
>Doha Qatar
>http://www.hatif.com
>
>
>
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