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 <[email protected]> # # 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>
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