404 Not Found error with Sipp UAC

classic Classic list List threaded Threaded
3 messages Options
Reply | Threaded
Open this post in threaded view
|

404 Not Found error with Sipp UAC

Steve Mitchell
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 <[hidden email]>
#
# 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
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Reply | Threaded
Open this post in threaded view
|

Re: 404 Not Found error with Sipp UAC

Ali Pey
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 <[hidden email]> 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 <[hidden email]>
#
# 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
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Reply | Threaded
Open this post in threaded view
|

Re: 404 Not Found error with Sipp UAC

Steve Mitchell
Thanks Ali! That's perfect.

Steve

On Fri, Nov 9, 2012 at 8:50 AM, Ali Pey <[hidden email]> wrote:
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 <[hidden email]> 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 <[hidden email]>
#
# 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
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users