Re: [Serusers] Problem with forwarding to a PSTN-Gateway :-(

Nobody can help me ? Or nobody wants to help me ? ;-)

I've made one change yesterday. I've configured the MediaProxy for me and deactivated the RTPProxy. The Calls between VoIP-and-VoIP go correctly...

It is just the Problem with the PSTN-Gateway :-(

Thanks

Dirk

Dirk Willbrandt wrote:

Hi all...

now i have the following scenario:

I want to forward calls with an 8 as prefix to a PSTN Gateway but when i place a call to i.e. 8004912345678 i get a busy-tone on my VoIP Phone and the call gets canceled.

My ser.cfg is configured out, so that i can say "theoretly it have to do" :-)

I've installed a RTPProxy too. I've took the rtpproxy from ser-cvs.

The connection between ser and rtp is ethablished when i start the ser.

On the PSTN-Gateway i can see, when i place a call, that a request comes to the Gateway but it isn't the IP-Address of my ser but the IP-Address of my VoIP Phone and this IP-Address isn't allowed to connect to the PSTN-Gateway.

Also, when i reconfigure the PSTN-Gateway so that the Phone-IP-Address is allowed to connect then i get a busy-tone on my phone, but the PSTN Phone rings. When i took the PSTN-Phone i hear nothing and my VoIP-Phone doesn't ring again.

Can anyone help me ? :-( I become crazy with it - i work now 2 weeks on this Problem :-(

Here is my ser.cfg for reference:

---

# # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $ # # simple quick-start config script #

# ----------- global configuration parameters ------------------------

#debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E)

# Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes #

check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 #children=4 #fifo="/tmp/ser_fifo"

alias="terralink.de" # myself=terralink.de alias="siiip.terralink.de" # myself=siiip.terralink.de alias="217.9.16.13" # myself

# ------------------ module loading ----------------------------------

# Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so"

loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so"

# Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so"

#loadmodule "/usr/local/lib/ser/modules/acc.so"

loadmodule "/usr/local/lib/ser/modules/nathelper.so"

loadmodule "/usr/local/lib/ser/modules/xlog.so"

# ----------------- setting module-specific parameters ---------------

# -- usrloc params -- modparam("usrloc", "db_url", "mysql://ser:ser@localhost/ser") #modparam("usrloc", "db_mode", 0) modparam("usrloc", "db_mode", 2)

# -- auth params -- #modparam("auth_db", "db_url", "sql://ser:ser@localhost/ser") #modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "password_column", "password") modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf")

# -- rr params -- modparam("rr", "enable_full_lr", 1)

# -- acc params -- #modparam("acc", "log_level", 1) #modparam("acc", "log_flag", 1 ) #modparam("acc", "log_missed_flag", 3) #modparam("acc", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") #modparam("acc", "radius_flag", 1) #modparam("acc", "radius_missed_flag", 3)

# -- nat params -- modparam("nathelper", "natping_interval", 10) modparam("nathelper","rtpproxy_sock", "/var/run/rtpproxy.sock")

# ------------------------- request routing logic -------------------

# main routing logic

route{ # zu viele Hops ? if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; # nachricht zu lang ? if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; };

   # we record-route all messages -- to make sure that
   # subsequent messages will go through our proxy; that's
   # particularly good if upstream and downstream entities
   # use different transport protocol
   record_route();

   # loose-route processing
   if (loose_route()) { t_relay(); break; };

   # labeled all transaction for accounting
   setflag(1);

   #if (!lookup("location")) {
   #        # call invitations to off-line users are reported 

using the # # acc_request action; to avoid duplicate reports on request # # retransmissions, request is processed statefuly (t_newtran, # # t_reply) # if ((method=="INVITE" || method=="ACK") && t_newtran() ) { # t_reply("404", "Not Found"); # #acc_request("404 Not Found"); # break; # }; # # all other requests to off-line users are simply replied # # statelessly and no reports are issued # #sl_send_reply("404", "Not Found"); # #break; #} else { # # user on-line; report on failed transactions; mark the # # transaction for reporting using the same number as # # configured above; if the call is really missed, a report # # will be issued # setflag(3); # # forward to user's current destination # t_relay(); # break; #};

   # if the request is for other domain use UsrLoc
   # (in case, it does not work, use the following command
   # with proper names and addresses in it)
   if (uri==myself) {
           rewritehost("siiip.terralink.de");
           #xlog("L_DBG", "time [%Tf] method <%rm> r-uri <%ru> 

from <%fu> contact <%ct>\n"); if (method=="REGISTER") { route(1); break; }; if (method=="INVITE") { fix_nated_contact(); record_route(); force_rtp_proxy(); if (uri=~"^sip:(.+)?@(.+)?") { route(3); break; } #else { break; }; }

           lookup("aliases");

           # native SIP destinations are handled using our USRLOC DB
           if (!lookup("location")) {
           #if (!lookup("location") || !lookup("aliases")) {
                   sl_send_reply("404", "Not Found");
                   break;
           };
   };
   # forward to current uri now; use stateful forwarding; that
   # works reliably even if we forward from TCP to UDP
   if (!t_relay()) {
           sl_reply_error();
   };

}

route[1] { #xlog("L_DBG", "Hier registriert sich jemand !\n"); if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }

#route[2] { #xlog("L_DBG", "Hier will jemand intern telefonieren !\n"); #}

route[3] { #xlog("L_DBG", "Hier will jemand extern telefonieren !\n"); #strip(1); #rewritehostport("217.9.16.13:5060"); rewritehostport("217.9.21.6:5060"); #forward( 217.9.16.11, 5060 ); append_branch("sip:sip@217.9.16.13"); #t_relay_to_udp("217.9.21.6", "5060"); if (!t_relay_to_udp("217.9.21.6", "5060")) { sl_reply_error(); break; }; }

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