26 Sep
2003
26 Sep
'03
8:04 p.m.
Greg,
You can set a timer when the phone rings and isn't answered.
modparam("tm", "fr_inv_timer", 20 )
20 is the time before the failure. You can change this.
Stephen
-----Original Message-----
From: Greg Fausak [mailto:greg@august.net]
Sent: Friday, September 26, 2003 11:56 AM
To: 'Jiri Kuthan'; 'Gavin Bensom'; serusers(a)lists.iptel.org
Subject: RE: [Serusers] voicemail question - dial plan example -
pstngatewayexample
Jiri,
I believe this is what you and I talked about
at VON. I am very interested in an easy way to
do serial forking. For instance, if I have 3 sip endpoints
like :
1000(a)myproxy.com
1010(a)myproxy.com
1020(a)myproxy.com
And finally, a voice mail reference:
1000(a)myvoicemailmediaserver.com
I currently am doing a serial forking like this:
First, create a table called 'forward' that works exactly like
'aliases'.
Then, put all 'forward' instructions in the table. This includes
forwarding
to voicemail server.
{
...
if(lookup("forward"))
{
xlog("L_WARN", "HUNTGROUP: %Ts call-id:%ci cseq:%ci\n");
t_on_failure("1");
}
if(!t_relay())
{
sl_reply_error();
break;
}
...
}
# here is the 'failure_route' for number forwarding
failure_route[1]
{
if(lookup("forward"))
{
xlog("L_WARN", "CONTINUE HUNT: %Ts call-id:%ci cseq:%ci
messageid:%mi\n");
t_on_failure("1");
append_branch();
t_relay();
break;
};
xlog("L_WARN", "NO CONTINUE HUNT: %Ts call-id:%ci cseq:%ci
messageid:%mi\n");
#sl_reply_error();
t_reply("486","Busy puppy.");
break;
}
This setup sort of works, but, it seems that I get many
t_relay() events that just hang and eventually time out,
which I don't understand how is happening.
Also, it would be nice if I could set a timer, say for 15 seconds, that
causes a failure to occur in the event a phone rings but isn't
answered.
---greg
There is other problem though -- the failure_route logic for
VM should be different
from that in route block -- you can't create transaction
state two times
(first time with t_relay, second time with t_newtran). SER
will certainly
complain with some error message in your logs.
I will introduce some new command which hides details of
transaction processing
from script writers and can be used from both route[] and
failure_route[].
(I didn't think of it before since I used voicemail on some
other machine
than proxy server.)
-jiri
if(uri=~"^sip:650[0-9]{7}@(mydomain\.com|10\.10\.10\.49)"){
if(uri=~"^sip:[0-9]{10}@(mydomain\.com|10\.10\.10\.49)"){
if(uri=~"^sip:1[0-9]{10}@(mydomain\.com|10\.10\.10\.49)"){
> rewritehostport("10.10.10.5:5060");
> log(1,"10 digit exp match w/leading 1");
> route(2);
> break;
> };
> #forward international N digit requests to gateway
if(uri=~"^sip:011[0-9]+@(mydomain\.com|10\.10\.10\.49)"){
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Thanks,
G.
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#
# $Id: ser.cfg,v 1.20 2003/05/31 21:12:19 jiri 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=8
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"
sip_warning=no
#
# ------------------ 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/vm.so"
loadmodule "/usr/local/lib/ser/modules/pa.so"
loadmodule "/usr/local/lib/ser/modules/msilo.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
#
#loadmodule "/usr/local/lib/ser/modules/nathelper.so"
#loadmodule "/usr/local/lib/ser/modules/uri.so"
#loadmodule "/usr/local/lib/ser/modules/group.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_db.so"
#
# ----------------- setting module-specific parameters
---------------
#
# -- usrloc params --
#
#modparam("usrloc", "db_mode", 0)
#
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
#
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in
this
config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
#
#
modparam("acc", "log_level", 1)
modparam("acc", "log_flag", 2)
modparam("acc", "log_missed_flag", 2)
modparam("acc", "log_fmt", "fimos")
#
#modparam("tm", "fr_inv_timer", 35) #INVITE timeout
#modparam("tm", "fr_timer", 30) #negative INVITE replay or no
#final reply for a request for ACK
#
modparam("voicemail", "db_url",
"sql://ser:heslo@localhost/ser")
#
#modparam("acc", "db_url", "sql://ser:heslo@localhost/ser")
#modparam("acc", "db_flag", 2)
#modparam("acc", "db_missed_flag", 2)
#
# ------------------------- request routing logic
-------------------
#
# main routing logic
#
alias=10.10.10.49 #sip server IP address
alias=jiffypop #sip server name
alias=mydomain.com #sip domain/realm
alias=jiffypop.mydomain.com #sip server FQDN
#
route{
log(1,"entering main route");
#prevent strangers from claiming to belong to our domain;
#if sender claims to be in our domain in From header field,
#better authenticate him
# code not inserted yet :)
# 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 (len_gt( 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;
};
setflag(2); #set flag for accounting
# 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=="REGISTER") {
# digest authentication
log(1,"request for registration");
if (!www_authorize("mydomain.com",
"subscriber")) {
www_challenge("mydomain.com", "0");
break;
};
save("location");
break;
};
/* ********** Dial out to PSTN logic ************* */
#forward numerical 7 digit requests to gateway
if(uri=~"^sip:[0-9]{7}@(mydomain\.com|10\.10\.10\.49)"){
#
append_hf("Remote-Party-ID:<sip:6508466600@mydomain.com>;party
=calling;id-type=subscriber;privacy=off;screen=no;\r\n");
rewritehostport("10.10.10.5:5060");
log(1,"7 digit expression match");
route(2);
break;
};
# strip 650 and forward to GW is user dials
650 before phone no.
strip(3);
rewritehostport("10.10.10.5:5060");
log(1,"650 area code dialed, 650 stripped");
route(2);
break;
};
#forward numerical 10 digit requests to
gateway, append a 1 first
prefix("1");
rewritehostport("10.10.10.5:5060");
log(1,"10 digit expression match, prefix 1");
route(2);
break;
};
#forward numerical 11 digit requests that
start with a 1 to GW
rewritehostport("10.10.10.5:5060");
log(1,"international expression match");
route(2);
break;
};
/* ********** VOICEMAIL logic ************* */
if (uri=~"^sip:voicemail\+@"){
t_newtran();
t_reply("100", "Trying -- just a second");
if(!vm("/tmp/am_fifo","announcement")){
t_reply("500", "SEMS error");
};
log(1,"sip:voicemail uri match");
break;
};
/* ****** Find Aliases and Locations of users ********* */
# It is very important to lookup "aliases" before looking up
"locations"
if(!lookup("aliases")){
log(1,"Couldn't find any matching alias");
sl_send_reply("404", "User does not exist");
break;
};
if(!lookup("location")) {
log(1,"unable to locate user");
route(3);
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
t_on_failure("1");
if (!t_relay()) {
sl_reply_error();
};
log(1,"eof");
}
route[2]{
log(1,"route[2]:SIP-to-PSTN call routed");
if(!t_relay()){
sl_reply_error();
};
}
route[3]{
log(1,"route[3]: voicemail processing");
if(method=="INVITE" || method=="ACK"){
log(1,"1st if entered in route[3] *vm*");
t_newtran();
t_reply("100","Trying -- just a second");
if(!vm("/tmp/am_fifo","voicemail")){ #
"announcement" was replaced with "voicemail"
log(1,"vm module
called and failed");
t_reply("500", "SEMS error");
};
break;
};
log(1,"end of route[3], 1st if check not entered");
}
failure_route[1]{
log(1,"failure_route[1]:jump to vm processing");
route(3);
# break;
}
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--
Jiri Kuthan http://iptel.org/~jiri/
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