[Serusers] Problem with forwarding to a PSTN-Gateway :-(
Dirk Willbrandt
dw at terralink.de
Wed Mar 30 16:49:22 CEST 2005
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 at 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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