[OpenSIPS-Users] 404 Not Found error with Sipp UAC

Ali Pey alipey at gmail.com
Fri Nov 9 16:50:42 CET 2012


Hi Steve,

The request-uri in your invite is to "service". In your opensips script, it
does a lookup on user and I don't think you have the user "service"
registered.

Either have sipp to register first, or in your opensips.cfg, put a
condition for calls from sipp to go to route one directly with no lookup.

Hope this is clear enough.

Regards,
Ali Pey


On Thu, Nov 8, 2012 at 8:05 PM, Steve Mitchell <swmitchell at gmail.com> wrote:

> 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>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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