[OpenSIPS-Users] 404 Not Found error with Sipp UAC
Steve Mitchell
swmitchell at gmail.com
Fri Nov 9 02:05:15 CET 2012
Hi,
I'm trying to get a simple scenario working to generate CDRs in bulk and am
using a basic configuration (generated with osipconfig) with the sipp UAC.
However, I continue to get a 404 Not Found response to the INVITE. My
config and UAC file are below.
Any thoughts?
Thanks much!
Steve
#
# $Id: opensips_residential.m4 9042 2012-05-17 13:57:10Z vladut-paiu $
#
# OpenSIPS residential configuration script
# by OpenSIPS Solutions <team at opensips-solutions.com>
#
# This script was generated via "make menuconfig", from
# the "Residential" scenario.
# You can enable / disable more features / functionalities by
# re-generating the scenario with different options.#
#
# Please refer to the Core CookBook at:
# http://www.opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#
####### Global Parameters #########
debug=6
fork=no
log_stderror=yes
log_facility=LOG_LOCAL1
#fork=yes
#children=4
/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes
/* uncomment the next line to enable the auto temporary blacklisting of
not available destinations (default disabled) */
#disable_dns_blacklist=no
/* uncomment the next line to enable IPv6 lookup after IPv4 dns
lookup failures (default disabled) */
#dns_try_ipv6=yes
/* comment the next line to enable the auto discovery of local aliases
based on revers DNS on IPs */
auto_aliases=no
listen=udp:10.145.185.49:5060 # CUSTOMIZE ME
disable_tcp=no
listen=tcp:10.145.185.49:5060 # CUSTOMIZE ME
disable_tls=yes
####### Modules Section ########
#set module path
mpath="/usr/local/opensips_proxy/lib64/opensips/modules/"
#### SIGNALING module
loadmodule "signaling.so"
#### StateLess module
loadmodule "sl.so"
#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timer", 5)
modparam("tm", "fr_inv_timer", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)
#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)
#### MAX ForWarD module
loadmodule "maxfwd.so"
#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"
#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)
#### URI module
loadmodule "uri.so"
modparam("uri", "use_uri_table", 0)
#### MYSQL module
loadmodule "db_mysql.so"
#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", 10)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
#### REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", 7)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
modparam("acc", "failed_transaction_flag", 3)
/* account triggers (flags) */
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)
modparam("acc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
####### Routing Logic ########
# main request routing logic
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
} else if (is_method("INVITE")) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
}
# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(1);
} else {
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK
after
# a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction ->
# ignore and discard
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
if ( !(is_method("REGISTER") ) ) {
if (from_uri==myself)
{
} else {
# if caller is not local, then called number must be local
if (!uri==myself) {
send_reply("403","Rely forbidden");
exit;
}
}
}
# preloaded route checking
if (loose_route()) {
xlog("L_ERR",
"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
if (!is_method("ACK"))
sl_send_reply("403","Preload Route denied");
exit;
}
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
# account only INVITEs
if (is_method("INVITE")) {
setflag(1); # do accounting
}
if (!uri==myself) {
append_hf("P-hint: outbound\r\n");
route(1);
}
# requests for my domain
if (is_method("PUBLISH|SUBSCRIBE"))
{
sl_send_reply("503", "Service Unavailable");
exit;
}
if (is_method("REGISTER"))
{
if ( proto==TCP || 0 ) setflag(7);
if (!save("location"))
sl_reply_error();
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# do lookup with method filtering
if (!lookup("location","m")) {
t_newtran();
t_reply("404", "Not Found");
exit;
}
# when routing via usrloc, log the missed calls also
setflag(2);
route(1);
}
route[1] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("2");
t_on_reply("2");
t_on_failure("1");
}
if (!t_relay()) {
send_reply("500","Internal Error");
};
exit;
}
branch_route[2] {
xlog("new branch at $ru\n");
}
onreply_route[2] {
xlog("incoming reply\n");
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
# uncomment the following lines if you want to block client
# redirect based on 3xx replies.
##if (t_check_status("3[0-9][0-9]")) {
##t_reply("404","Not found");
## exit;
##}
}
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword.
-->
<send retrans="1000">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true" rrs="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK [next_url] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
[routes]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause milliseconds="10000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="1000">
<![CDATA[
BYE [next_url] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
[routes]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="500, 1000, 1500, 2000"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="500"/>
</scenario>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20121108/d9f13745/attachment-0001.htm>
More information about the Users
mailing list