Hi guys,
I'm new to Asterisk, so I'm hoping someone can guide me :-)
Currently, I am having the configuration as follows :
PSTN -> Cisco router -> Sip Express Router -> Asterisk Voicemail
I'm able to get the part from PSTN to Sip Express Router working, but
I can't integrate Asterisk with Sip Express Router (SER).
Basically, SER does all the registering and forwarding of calls. I
need to implement the voicemail in Asterisk, whereby a user calls a
certain IP Phone, and if the user does not pick up the call in time,
the call is diverted to Asterisk's voicemail.
However, I am unable to get Asterisk to activate the voicemail upon
missed calls. Please kindly advise.
Regards,
YY
My current settings are as follows :
-------------
------------
SER
------------
-------------
1. ser.cfg (SER's config file)
-----------------------------------------
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
# Uncomment these lines to enter debugging mode
debug=3
fork=yes
listen=202.122.25.106
log_stderror=yes
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
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 --
# store user location in memory, not using database
modparam("usrloc", "db_mode", 0)
modparam("rr", "enable_full_lr", 1)
# -- tm params --
# set time for which ser will be waiting for a final response;
# fr_inv_timer sets value for INVITE transactions,
# fr_timer for all others
modparam("tm","fr_inv_timer",15)
# ------------------------- 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 > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
setflag(1);
# 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();
};
# loose-route processing
if (loose_route()) {
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){
route(1);
break;
};
if (uri==myself) {
if (method=="REGISTER") {
route(2);
break;
};
setflag(4);
# attempt handoff to PSTN
if (uri=~"^sip:[EMAIL PROTECTED]") { ## This assumes that
the caller is
log(1, "Forwarding to PSTN"); ##
registered in our realm
forward(10.10.10.3, 5060); ## Our
Cisco router
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
#acc_rad_request("404");
break;
};
# timeout occurred ... now to forward to Asterisk's voicemail
service
if(method == "INVITE" && isflagset(4)) {
t_on_failure("1");
};
};
route(1);
}
# -------------------------------
# Route Processing
# -------------------------------
route[1]{
if(!t_relay()){
sl_reply_error();
};
}
route[2]{
if(!save("location")){
sl_reply_error();
}
}
# voicemail activation!!
#
failure_route[1] {
log(1,"Activating voicemail!!\n");
forward(202.122.25.106, 5061);
}
---------------------------
--------
--------
ASTERISK
--------
--------
voicemail.conf
---------------
[default]
1012 => 1234, YY, [EMAIL PROTECTED]
sip.conf
-----------
port=5061 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
[1012]
type=friend
username=1012
insecure=yes
canreinvite=no
context=test
mailbox=1012
host=202.122.25.106
nat=no
extensions.conf
----------------
[test]
;leave voice messages
exten => 1012, 1, Wait(1)
exten => 1012, 2, VoiceMail(u1012)
exten => 1012, 3, Hangup
;play voice messages
exten => 2012, 1, Wait(1)
exten => 2012, 2, VoiceMailMain()
exten => 2012, 3, Hangup
------------------------------
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