Hi Henning,
There is no error. Just it doesn't behave the way it should. By the way the thread you
replied to has an error in the config I had sent. Please find it corrected below. Thank
you
#------CONFIG BEGINS------------------
mpath="/lib/kamailio/modules_k/"
debug=3
fork=yes
children=4
auto_aliases=no
alias=localhost
alias=192.168.10.1
alias=192.168.10.2
alias=192.168.10.3
alias=192.168.10.4
alias=192.168.10.5
alias=192.168.10.6
disable_tcp=yes
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "/lib/kamailio/modules/tm.so"
loadmodule "textops.so"
modparam("rr", "enable_full_lr", 1)
route {
# Sanity Check
# ------------
# filter too old messages
if (!mf_process_maxfwd_header("10")) {
log("LOG: Too many hops\n");
sl_send_reply("483","Too Many Hops");
break;
};
if(msg:len>2048) {
sl_send_reply("413", "message too large to be forwarded over
UDP without fragmentation");
exit;
};
# Record Route
# --------------
if (method != "REGISTER") {
record_route();
};
# Loose Route
# -----------
if (loose_route()) {
route(1);
return;
};
# Call Type Processing
# --------------------
if (uri != myself) {
route(1);
return;
};
if (uri == myself) {
if (method == "BYE") {
route(4);
return;
} else if (method == "CANCEL") {
route(4);
return;
} else if (method == "INVITE") {
route(3);
return;
} else if (method == "NOTIFY") {
sl_send_reply("200", "Understood");
return;
} else if (method == "OPTIONS") {
sl_send_reply("200", "Got it");
return;
}
};
route(1);
}
# Default Message Handling
# -----------------------
route[1] {
t_on_reply("1");
if (!t_relay()) {
sl_reply_error();
};
}
# INVITE Message Handling
# ----------------------------------
# ----------------------------------
route[3] {
if (uri =~ "^sip:011[0-9]@*") {
rewritehostport("sip.voipprovider.com:5060");
if (search("^(Contact|m): .*user01*(a)(127\.0\.0\.1|localhost)"))
{
force_send_socket(192.168.10.2:5060);
};
route(1);
return;
};
}
# CANCEL and BYE Message Handling
# ----------------------------------
route[4] {
route(1);
}
Cheers
--- On Tue, 2/23/10, Henning Westerholt <henning.westerholt(a)1und1.de> wrote:
From: Henning Westerholt
<henning.westerholt(a)1und1.de>
Subject: Re: [Kamailio-Users] Forward calls from Asterisk to SIP provider via Kamailio
for termination
To: users(a)lists.kamailio.org
Cc: "Slot Zero" <slotzero1(a)yahoo.com>
Date: Tuesday, February 23, 2010, 7:57 AM
On Saturday 20 February 2010, Slot
Zero wrote:
I am a Kamailio noob :). I am trying to get
Asterisk
to forward calls to
my SIP provider via Kamailio.
The same machine is running Kamailio and
Asterisk. I do not want to consume credentials as they
have to be passed
on all the way to my SIP provider. There is no
NAT of
any sorts. SIP
Phone/Users connect to Asterisk. I do not need to
authenticate when
forwarding call from Asterisk to Kamailio as
they are both running on the
same server but I do need to make sure that
Kamailio dials and forwards
011+number to be sent from local host port
5062(Asterisk listener) to SIP provider only.
I have 6 Public IP addresses
mapped on the server. I want to use the
force_send_socket to allow me to
change source IP of SIP requests when being sent
to
the SIP provider on the
basis of credentials username in the request. I
have pasted my config
below. Please tell me what am I doing wrong
here. In the kamctlrc file i
have SIP_DOMAIN=localhost
Hi Slot,
do you observe an error with your quoted configuration, or
it does not behave
like you expect?
Cheers,
Henning