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...@*") {
rewritehostport("sip.voipprovider.com:5060");
if (search("^(Contact|m): .*user01*@(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.westerh...@1und1.de> wrote:
> From: Henning Westerholt <henning.westerh...@1und1.de>
> Subject: Re: [Kamailio-Users] Forward calls from Asterisk to SIP provider via
> Kamailio for termination
> To: users@lists.kamailio.org
> Cc: "Slot Zero" <slotze...@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
>
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