[Serusers] [about SER]networking among three SIP servers.....

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15

Dear........

I want to networking among three SIP servers. But now I am trying using two servers likes three servers. That is,

- source : computer A ip address ; 192.168.178.194 sipuri : sip:mj@192.168.178.194

- via : computer B ip address : 192.168.178.193 sipuri : sip:hj@192.168.178.193

- destination : computer A ip address :192.168.178.194 sipuri : sip:mj@192.168.178.194

And I am using sipp to test (I modified the "via" part based on original uac module).

=====================================================

<< computer B's test.cfg >>

# # $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $ # # welcome message for new subscribers; based on exec #

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

debug=3 # debug level (cmd line: -dddddddddd) # must be yes since REGISTER processing causes an INVITE to be sent, # which needs to be processed by another process fork=yes children=4

alias="kims.myserver.sampro.cse" alias="myserver.sampro.cse" alias="192.168.178.193"

# debugging log_stderror=yes # (cmd line: -E)

mhomed=yes

fifo="/tmp/ser_fifo"

# ------------------ 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"

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

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

# -- usrloc params --

modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" ) modparam("usrloc", "db_url", "mysql://root:heslo@localhost/ser")

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

# main routing logic

route{ log(1, "\n\n"); log(1, "START!!!\n"); # 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>=204822) { 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=="INVITE") record_route(); # 1=loose routing if(method=="INVITE"){ log(1, "receive INVITE\n"); record_route(); sl_send_reply("200", "ok"); };

# loose-route processing if (loose_route()) { t_relay(); break; }; # if(method=="ACK"){ # log(1, "receive ACK\n"); # };

log(1, "record-route processing completed\n");

# 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(uri=="sip:mijin@sam.cse"){ # forward(192.168.178.194, 5060); # };

if (uri=~"sip:192.168.178.193"){

if (method=="REGISTER") {

log(1, "receive REGISTER\n"); # save location before initiating welcome save("location"); log(1, "save &#236;&#8222;±&#234;³&#181;\n"); record_route(); sl_send_reply("200", "ok"); # welcome message log(1, "route3&#236;œ¼&#235;¡œ &#236;&#157;´&#235;&#143;™ \n"); route(3); break; };

# native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "Usrloc DB&#236;&#8212;&#144;&#236;&#8222;œ &#236;°¾ &#236;&#157;&#8222; &#236;&#710;&#732; &#236;&#8212;†&#236;&#157;Œ \n"); break; }; }; if(method=="BYE"){ log(1, "receive BYE\n"); record_route(); sl_send_reply("200", "ok"); break; }; t_relay(); }

# welcome message -- if a REGISTER succeeded, look if it is the first-time; # if so, initiate click-to-dial transaction to set up call to an announcement # server; edit the config values first to make it work route[3] {

if (!exec_msg(' # config: # --announcement server URI ANS="sip:hjkim@myserver.sampro.cse" # --SIP domain # DOMAIN="192.168.178.193" DOMAIN="192.168.178.193" # ctd CTD=${HOME}/sip_router/examples/ctd.sh # ------------------------------------ # check if first time ... SIP_UID=`echo $SIP_HF_TO | sed -e "s/^.*sip:([a-zA-Z0-9_.]*)@.*$/\1/g"` QUERY="select flag from subscriber where username="$SIP_UID"; update subscriber set flag="x" where username="$SIP_UID" "; mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null # ... if so, c-t-d to announcement server if [ "$?" -ne 0 ] ; then # flag was not set to x yet -- first-time registration; # initiate a call from telephone of the user to an announcement server $CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1 fi ')) { log(1, "welcome exec failed\n"); }

log(1, "-------route3 &#235;&#129;&#157; \n"); }

As you see, I added

modparam("rr", "enable_full_lr", no)

if(uri=="sip:mijin@192.168.178.194"){ forward(192.168.178.194, 5060); append_branch("sip:hjkim@192.168.178.193:5060"); log(1, "next hop...\n"); break; };

but, I received error message in computer B(via):

8(12253) ERROR: forward_reply: no 2nd via found in reply

What does it mean? What can I do for doing what I want ??? Is there any problem in my test.cfg file ?????

I would like to know in detail . please tell me what to do...............................

Song

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