<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2800.1400" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT size=2>I am using asterisk as my voicemail system and ser as my sip
server. It works fine for a specific called number(4243) by using
append_branch() function. But I don't know how to setup ser to forward any
called number to asterisk if no one answer the phone. I mean something
like forward() function so I only need to specify the IP and port of
asterisk. </FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>Here is my ser.cfg:</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>#<BR># $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp
$<BR>#<BR># simple quick-start config script<BR>#</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2># ----------- global configuration parameters
------------------------</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2>#debug=3 #
debug level (cmd line: -dddddddddd)<BR>#fork=yes<BR>#log_stderror=no # (cmd
line: -E)</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2>/* Uncomment these lines to enter debugging mode
<BR>fork=no<BR>log_stderror=yes<BR>*/<BR>debug=7</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2>check_via=no # (cmd. line:
-v)<BR>dns=no #
(cmd. line: -r)<BR>rev_dns=no # (cmd. line:
-R)<BR>#port=5060<BR>#children=4<BR>fifo="/tmp/ser_fifo"</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2># ------------------ module loading
----------------------------------</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2># Uncomment this if you want to use SQL database<BR>loadmodule
"/usr/lib/ser/modules/mysql.so"</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2>loadmodule "/usr/lib/ser/modules/sl.so"<BR>loadmodule
"/usr/lib/ser/modules/tm.so"<BR>loadmodule
"/usr/lib/ser/modules/rr.so"<BR>loadmodule
"/usr/lib/ser/modules/maxfwd.so"<BR>loadmodule
"/usr/lib/ser/modules/usrloc.so"<BR>loadmodule
"/usr/lib/ser/modules/registrar.so"</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2># Uncomment this if you want digest authentication<BR>#
mysql.so must be loaded !<BR>loadmodule
"/usr/lib/ser/modules/auth.so"<BR>loadmodule
"/usr/lib/ser/modules/auth_db.so"</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2># ----------------- setting module-specific parameters
---------------</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2># -- usrloc params --</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2>modparam("usrloc", "db_mode", 0)</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2># Uncomment this if you want to use SQL database <BR># for
persistent storage and comment the previous line<BR>modparam("usrloc",
"db_mode", 2)</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2># -- auth params --<BR># Uncomment if you are using auth
module<BR>#<BR>modparam("auth_db", "calculate_ha1", yes)<BR>#<BR># If you set
"calculate_ha1" parameter to yes (which true in this config), <BR># uncomment
also the following parameter)<BR>#<BR>modparam("auth_db", "password_column",
"password")</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2># -- rr params --<BR># add value to ;lr param to make some
broken UAs happy<BR>modparam("rr", "enable_full_lr", 1)</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2>modparam("tm", "fr_inv_timer", 15)<BR>modparam("tm",
"fr_timer", 10)</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2># ------------------------- request routing logic
-------------------</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2># main routing logic</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2>route{</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2> # initial sanity checks -- messages with<BR> #
max_forwards==0, or excessively long requests<BR> if
(!mf_process_maxfwd_header("10")) {<BR> sl_send_reply("483","Too Many
Hops");<BR> break;<BR> };<BR> if ( msg:len > max_len )
{<BR> sl_send_reply("513", "Message too
big");<BR> break;<BR> };</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2> # we record-route all messages -- to make sure
that<BR> # subsequent messages will go through our proxy; that's<BR> #
particularly good if upstream and downstream entities<BR> # use different
transport protocol<BR> record_route(); <BR> # loose-route
processing<BR> if (loose_route())
{<BR> t_relay();<BR> break;<BR> };</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2> # if the request is for other domain use
UsrLoc<BR> # (in case, it does not work, use the following
command<BR> # with proper names and addresses in it)<BR> if
(uri==myself) {</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2> if (method=="REGISTER") {</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2># Uncomment this if you want to use digest
authentication<BR> if (!www_authorize("seti", "subscriber"))
{<BR> www_challenge("seti",
"0");<BR> break;<BR> };</FONT></DIV>
<DIV> </DIV>
<DIV><FONT
size=2> save("location");<BR> break;<BR> };<BR></FONT></DIV>
<DIV><FONT size=2> # native SIP destinations are handled using our
USRLOC DB<BR> if (!lookup("location")) {<BR> #Handle PSTN
calls.<BR> if (uri=~"^sip:8500@.*") #To asterisk
voicemail
admin.<BR> {<BR> record_route();<BR> rewritehostport("Asterisk
server IP:PORT");<BR> forward(<Asterisk server
IP:PORT>);<BR> }<BR> else<BR> {<BR> record_route();<BR> rewritehostport("PSTN
IP:PORT");<BR> forward(PSTN
IP:PORT);<BR> };<BR> };<BR> };<BR> t_on_failure("1");<BR> #
forward to current uri now; use stateful forwarding; that<BR> # works
reliably even if we forward from TCP to UDP<BR> if (!t_relay())
{<BR> sl_reply_error();<BR> };</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2>}<BR>failure_route[1]
{<BR> append_branch("sip:4243@AsteriskIP:Port");<BR> t_relay();<BR>}</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2></FONT> </DIV></BODY></HTML>