check out the gettingstarted callfwd-features config. You need to test
for status in your failure route.
g-)
ram wrote:
Hi all
Since long iam running with CDROUTER (0.814)
Now i have got another server for testing with new Version of SER
So i took the stable version from
onsip.org <http://onsip.org>
and copied all the users and config files from old server to new server
and modified accoring to the new setup
when the user not available or busy iam sending to voice mail (
asterisk another server)
but i have new problem here with new version of ser is
when ever i call from user X to user Y ( once its start ringing)
if i disconnect the call on X Side, its going to voice mail. ( and
other side iam able to see Y user connection hangup)
same setup working with my old server with out any problem,
and i ahve changed many things in the config, but no luck,
so posting in group for the suggestions.
OLD SERVER
---------------------
version: ser 0.8.12 (i386/linux)
flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.168 2003/10/12 15:09:08 andrei Exp $
main.c compiled on 19:40:54 Jul 19 2004 with gcc 3.2
-------------------------------------------
NEW SERVER
version: ser 0.9.7-pre1 (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST,
DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.197.2.1 <http://1.197.2.1> 2005/07/25 16:56:24
andrei Exp $
main.c compiled on 15:49:21 Nov 3 2006 with gcc 3.4.6
-------------------------------------
here is my config
# does the user wish redirection on no availability? (i.e., is he
# in the voicemail group?) -- determine it now and store it in
# flag 4, before we rewrite the flag using UsrLoc
if (is_user_in("Request-URI", "voicemail")) {
setflag(4);
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
# handle user which was not found
route(4);
break;
};
# if user is on-line and is in voicemail group, enable redirection
if (method == "INVITE" || method=="BYE" ||
method=="ACK" &&
isflagset(4)) {
setflag(1); # for accounting
t_on_failure("1");
};
t_relay();
}
# ------------- handling of unavailable user ------------------
route[3]{
log(1,"route[3]:no user location: foward to voicemail");
rewritehostport("asteriskip:5090");
t_relay();
}
break;
}
route[4] {
# non-Voip -- just send "off-line"
if (!(method == "INVITE" || method == "ACK" || method ==
"CANCEL")) {
sl_send_reply("404", "Not Found");
break;
};
# not voicemail subscriber
if (!isflagset(4)) {
sl_send_reply("404", "Not Found and no voicemail
turned on");
break;
};
# forward to voicemail now
rewritehostport("asteriskip:5090");
t_relay_to_udp("asteriskip", "5090");
}
failure_route[1] {
revert_uri();
rewritehostport("asteriskip:5090");
append_branch();
t_relay_to_udp("asteriskip", "5090");
}
any suggestion will be appriciated
Ram
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