Hi, I have an asterisk server running with an private IP. This asterisk forwards all calls to a SER server with a public IP. The SER server then forwards its calls to a public SIP provider. The problem now is that SER tries to stay in the loop which it shouldn't because there is no media proxy running. I don't get any audio because of this issue. But if I register the asterisk box directly to the SIP provider it works. Does anybody know how to fix this. My ser.cfg debug=4 # debug level (cmd line: -dddddddddd) #debug=0 fork=no log_stderror=yes # (cmd line: -E)

check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #listen=0.0.0.0 #listen=82.98.89.140 port=5060 children=4 fifo="/tmp/ser_fifo"

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

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

loadmodule "/usr/local/ser/lib/ser/modules/sl.so" loadmodule "/usr/local/ser/lib/ser/modules/tm.so" loadmodule "/usr/local/ser/lib/ser/modules/rr.so" loadmodule "/usr/local/ser/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/ser/lib/ser/modules/usrloc.so" loadmodule "/usr/local/ser/lib/ser/modules/registrar.so" loadmodule "/usr/local/ser/lib/ser/modules/textops.so" loadmodule "/usr/local/ser/lib/ser/modules/avpops.so" #loadmodule "/usr/local/ser/lib/ser/modules/group.so" loadmodule "/usr/local/ser/lib/ser/modules/xlog.so" loadmodule "/usr/local/ser/lib/ser/modules/auth.so" loadmodule "/usr/local/ser/lib/ser/modules/nathelper.so" loadmodule "/usr/local/ser/lib/ser/modules/uri.so"

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

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

# -- usrloc params --

modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1)

#modparam("registrar", "nat_flag", 6) #modparam("nathelper", "natping_interval", 30) # Ping interval 30 s #modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT

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

# main routing logic

route{

# initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { 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 if (!method=="REGISTER") record_route();

# subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; };

if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); 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) { if (method=="ACK") { route(1); break; }

if (method=="REGISTER") { #record_route(); save("location"); break; }; if (method=="INVITE") { #if (uri =~ "sip:[0-9]@*") { # if (nat_uac_test("19")) { # fix_nated_contact(); # fix_nated_sdp("3"); # } # route(3); # break; #} if (uri =~ "sip:[0-9]@*") { # record_route(); route(3); break; } };

lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; };

# native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); }

route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; }

route[3] { if (uri =~ "sip:[0-9]@*") { log(1, "Forwarding to mg3.net-m.de \n"); #rewritehostport("192.168.13.102:5060"); rewritehostport("62.214.145.199:5060"); #forward(62.214.145.199, 5060); route(1); break; } }

My extensions.conf [toser] exten => _X.,1,Dial(sip/${EXTEN}@82.98.89.139)

Thanks for any help Ciao Thorsten