[OpenSIPS-Users] [OpenSIPS] sample script that works with cdrtools, freeradius, nat, drouting

[Jinsong Hu]

Go ahead. I am not sure how good it is. if you find problems , please let me 
know.

I am sure there must be problems there. But since I am also new, so I can't 
spot them all. that is why I posted it.  it will be nice that there are 
people spot the problem
and tell me. I can continue to improve it. when it is good enough, maybe 
this can be put in opensips repository
for everybody to share.

Jinsong


----- Original Message ----- 
From: "Khan" <khansfriend at gmail.com>
To: "Jinsong Hu" <jinsong_hu at hotmail.com>
Cc: <users at lists.opensips.org>
Sent: Monday, May 11, 2009 11:55 AM
Subject: Re: [OpenSIPS-Users] [OpenSIPS] sample script that works with 
cdrtools, freeradius, nat, drouting


Thanks Jimmy,

I have been looking for some sample script for ages at every possible
place. So, far no luck but its really nice of you to post your script
here. I always wondered with open source software what is the deal of
posting your functional script (hiding your confidential info) but
never had an answer. Finally I see a brave person posting script...

Would you mind if i try your script and see if it works for me???


Thanks,




On Mon, May 11, 2009 at 1:31 AM, Jinsong Hu <jinsong_hu at hotmail.com> wrote:
> Hi, There:
> It looks ag-projects is maintaining the cdrtools, media proxy. but I
> searched around and didn't find anywhere there is a script that supports 
> all
> the needed feature: cdrtools, mediaproxy, nat_traversal, and drouting. so
> now I'm trying to be a little brave and post my script that includes all
> above. this script doesn't handle instance message, but only voice calls.
> can any body spot problems with this script ?
> The goal of the script is to let locally registered user to use gateway to
> make outgoing call, and receive incoming call. the numbering plan is for 
> US.
> free radius should have good authenticaing and accounting for different
> messages, and some special DID are mapped to several numbers and routed to
> asterisk. Hopefully this script will be useful for a general VOIP carrier.
> I try to paste the document to be comment. Hopefully, by going through
> this exercise, we can get a good starting script for people to use as a
> model starting script.
>
>
> Jimmy
>
>
>
> #######################################################################
> #
> # $Id: opensips.cfg,v 1.13 2009/05/11 06:06:00 jinsong Exp $
> #
> # OpenSIPS basic configuration script
> # by Anca Vamanu <anca at voice-system.ro>
> #
> # Please refer to the Core CookBook at
> http://www.opensips.org/dokuwiki/doku.php
> # for a explanation of possible statements, functions and parameters.
> #
> #INVITE :Invites a user to a call
> #ACK : Acknowledgement is used to facilitate reliable message exchange for
> INVITEs.
> #BYE :Terminates a connection between users
> #CANCEL :Terminates a request, or search, for a user. It is used if a 
> client
> sends an INVITE and then changes its decision to call the recipient.
> #OPTIONS :Solicits information about a server's capabilities.
> #REGISTER :Registers a user's current location
> #INFO :Used for mid-session signaling
> #MESSAGE : IMS send message
> #SUBSCRIBE : IMS presence subscribe message
> #PUBLISH: IMS publish message
>
> #1xx: Provisional -- request received, continuing to process the request;
> #2xx: Success -- the action was successfully received, understood, and
> accepted;
> #3xx: Redirection -- further action needs to be taken in order to complete
> the request;
> #4xx: Client Error -- the request contains bad syntax or cannot be 
> fulfilled
> at this server;
> #5xx: Server Error -- the server failed to fulfill an apparently valid
> request;
> #6xx: Global Failure -- the request cannot be fulfilled at any server.
>
> #This function sets the value of the flag given as parameter to 1 (true).
> The value of the parameter must be an integer between 0 and 31.
>
>
>
>
>
> ####### Global Parameters #########
>
> debug=3
> log_stderror=no
> log_facility=LOG_LOCAL0
>
> fork=yes
> children=4
>
> /* uncomment the following lines to enable debugging */
> #debug=6
> #fork=no
> #log_stderror=yes
>
> /* uncomment the next line to disable TCP (default on) */
> #disable_tcp=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
>
> #disable dns to scale
> dns=no
> rev_dns=no
>
> /* uncomment the next line to disable the auto discovery of local aliases
> based on revers DNS on IPs (default on) */
> #auto_aliases=no
> alias=machinename.somedomain.com
>
>
>
> /* uncomment the following lines to enable TLS support (default off) */
> #disable_tls = no
> #listen = tls:your_IP:5061
> #tls_verify_server = 1
> #tls_verify_client = 1
> #tls_require_client_certificate = 0
> #tls_method = TLSv1
> #tls_certificate = "/etc/opensips/tls/user/user-cert.pem"
> #tls_private_key = "/etc/opensips/tls/user/user-privkey.pem"
> #tls_ca_list = "/etc/opensips/tls/user/user-calist.pem"
>
>
> port=5060
>
> /* uncomment and configure the following line if you want opensips to
> bind on a specific interface/port/proto (default bind on all available)
> */
> #listen=udp:192.168.1.2:5060
>
>
> ####### Modules Section ########
>
> #set module path
> mpath="/usr/lib/opensips/modules/"
>
> /* uncomment next line for MySQL DB support */
> loadmodule "db_mysql.so"
> loadmodule "mi_fifo.so"
> loadmodule "sl.so"
> loadmodule "tm.so"
> loadmodule "rr.so"
> loadmodule "maxfwd.so"
> loadmodule "usrloc.so"
> loadmodule "signaling.so"
> loadmodule "registrar.so"
> loadmodule "textops.so"
> loadmodule "uri_db.so"
> loadmodule "uri.so"
> loadmodule "xlog.so"
> loadmodule "acc.so"
> /* uncomment next lines for MySQL based authentication support
> NOTE: a DB (like db_mysql) module must be also loaded */
> loadmodule "auth.so"
> loadmodule "auth_db.so"
> /* uncomment next line for aliases support
> NOTE: a DB (like db_mysql) module must be also loaded */
> loadmodule "alias_db.so"
> /* uncomment next line for multi-domain support
> NOTE: a DB (like db_mysql) module must be also loaded
> NOTE: be sure and enable multi-domain support in all used modules
> (see "multi-module params" section ) */
> loadmodule "domain.so"
> /* uncomment the next two lines for presence server support
> NOTE: a DB (like db_mysql) module must be also loaded */
> #loadmodule "presence.so"
> #loadmodule "presence_xml.so"
>
> #loadmodule "carrierroute.so"
> loadmodule "drouting.so"
> loadmodule "siptrace.so"
> loadmodule "pike.so"
> loadmodule "ratelimit.so"
>
> loadmodule "auth_radius.so"
> loadmodule "avp_radius.so"
> #loadmodule "uri_radius.so"
> loadmodule "group_radius.so"
>
> loadmodule "dispatcher.so"
>
> loadmodule "dialog.so"
> loadmodule "mediaproxy.so"
> #loadmodule "nathelper.so"
> loadmodule "nat_traversal.so"
>
> # ----------------- setting module-specific parameters ---------------
>
>
> # ----- mi_fifo params -----
> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
>
>
> # ----- rr params -----
> # add value to ;lr param to cope with most of the UAs
> modparam("rr", "enable_full_lr", 1)
>
>
> # ----- rr params -----
> #modparam("registrar", "method_filtering", 1)
> /* uncomment the next line to disable parallel forking via location */
> # modparam("registrar", "append_branches", 0)
> /* uncomment the next line not to allow more than 10 contacts per AOR */
> modparam("registrar", "max_contacts", 10)
>
>
> # ----- uri_db params -----
> /* by default we disable the DB support in the module as we do not need it
> in this configuration */
> modparam("uri_db", "use_uri_table", 0)
> modparam("uri_db", "db_url", "")
>
>
> # ----- acc params -----
> /* what sepcial events should be accounted ? */
> #modparam("acc", "early_media", 1)
> #modparam("acc", "report_ack", 1)
> #modparam("acc", "report_cancels", 1)
> /* by default ww 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)
> /* uncomment the following lines to enable DB accounting also */
> #modparam("acc", "db_flag", 1)
> #modparam("acc", "db_missed_flag", 1)
>
>
> # global acc parameters
> modparam("acc", "failed_transaction_flag", 1)
> modparam("acc", "report_cancels", 0)
> modparam("acc", "report_ack", 0)
> modparam("acc", "early_media", 0)
>
> modparam("acc", "log_level", 1)
> modparam("acc", "log_flag", 1)
> modparam("acc", "log_missed_flag", 1)
>
> modparam("acc|auth_radius|group_radius|avp_radius", "radius_config",
> "/etc//radiusclient-ng/radiusclient.conf")
> modparam("acc", "radius_flag", 1)
> modparam("acc", "radius_missed_flag", 1)
> modparam("acc", "radius_extra", "User-Name=$Au; \
> Calling-Station-Id=$from; \
> Called-Station-Id=$to; \
> Sip-Translated-Request-URI=$ru; \
> Sip-RPid=$avp(s:rpid); \
> Source-IP=$avp(s:source_ip); \
> Source-Port=$avp(s:source_port); \
> SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
> Canonical-URI=$avp(s:can_uri); \
> Billing-Party=$avp(s:billing_party);
> \
> Divert-Reason=$avp(s:divert_reason);
> \
> User-Agent=$hdr(user-agent); \
> Contact=$hdr(contact); \
> Event=$hdr(event); \
> ENUM-TLD=$avp(s:enum_tld)")
>
> modparam("siptrace", "db_url",
> "mysql://opensips:password@localhost/opensips")
> modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)")
> modparam("siptrace", "trace_on", 1)
> modparam("siptrace", "trace_flag", 2)
>
>
> # ----- usrloc params -----
> #0 - This disables database completely. Only memory will be used. Contacts
> will not survive restart.
> #1 - Write-Through scheme. All changes to usrloc are immediately reflected
> in database too.
> #2 - Write-Back scheme. All changes are made to memory and database
> synchronization is done in the timer.
> #3 - DB-Only scheme. No memory
> #modparam("usrloc", "db_mode", 0)
> /* uncomment the following lines if you want to enable DB persistency
> for location entries */
> modparam("usrloc", "db_mode", 2)
> modparam("usrloc", "db_url",
> "mysql://opensips:password@localhost/opensips")
>
>
> # ----- auth_db params -----
> /* uncomment the following lines if you want to enable the DB based
> authentication */
> modparam("auth_db", "calculate_ha1", yes)
> modparam("auth_db", "password_column", "password")
> modparam("auth_db", "db_url",
> "mysql://opensips:password@localhost/opensips")
> modparam("auth_db", "load_credentials", "")
>
>
> # ----- alias_db params -----
> /* uncomment the following lines if you want to enable the DB based
> aliases */
> #modparam("alias_db", "db_url",
> # "mysql://opensips:password@localhost/opensips")
>
>
> # ----- domain params -----
> /* uncomment the following lines to enable multi-domain detection
> support */
> #modparam("domain", "db_url",
> # "mysql://opensips:password@localhost/opensips")
> #modparam("domain", "db_mode", 1) # Use caching
>
>
> # ----- multi-module params -----
> /* uncomment the following line if you want to enable multi-domain support
> in the modules (dafault off) */
> #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1)
>
>
> # ----- presence params -----
> /* uncomment the following lines if you want to enable presence */
> #modparam("presence|presence_xml", "db_url",
> # "mysql://opensips:password@localhost/opensips")
> #modparam("presence_xml", "force_active", 1)
> #modparam("presence", "server_address", "sip:192.168.1.2:5060")
>
> # ----- carrierroute params -----
> /* uncomment the following line if you want to enable carrierroute support
> in the modules (dafault off) */
> #modparam("carrierroute", "db_url",
> "mysql://opensips:password@localhost/opensips")
> #modparam("carrierroute", "config_source", "db")
>
> modparam("drouting", "db_url",
> "mysql://opensips:password@localhost/opensips")
> modparam("drouting", "ruri_avp", '$avp(dr_ruri)')
>
> modparam("drouting", "config_source", "db")
>
> modparam("dispatcher", "db_url",
> "mysql://opensips:password@localhost/opensips")
>
> modparam("nat_traversal", "keepalive_state_file",
> "/var/run/opensips/keepalive_state")
>
> modparam("mediaproxy","mediaproxy_socket",
> "/var/run/mediaproxy/dispatcher.sock")
> modparam("mediaproxy", "mediaproxy_timeout", 500)
> modparam("mediaproxy", "signaling_ip_avp", "$avp(s:nat_ip)")
> modparam("mediaproxy", "media_relay_avp", "$avp(s:media_relay)")
>
>
>
> ####### Routing Logic ########
>
>
> # main request routing logic
>
> route{
>
> if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","Too Many Hops");
> exit;
> }
>
> if (msg:len >= 2048 ) {
> sl_send_reply("513", "Message too big");
> exit;
> };
>
> #pike_check_req Process the source IP of the current request and returns
> false if the IP was exceeding the blocking limit
> if (!pike_check_req()) { exit; };
>
> #rate limit
> if (is_method("INVITE|REGISTER|SUBSCRIBE")) {
> #rl_check The method will return an error code if the limit
> for the matched algorithm is reached.
> if (!rl_check()) {
> #For the current request, a "503 - Server
> Unavailable" reply is sent back.
> rl_drop();
> exit;
> };
> };
>
>
> #we only handle voice, not any other things.
> if (is_method("PUBLISH|MESSAGE|SUBSCRIBE"))
> {
> sl_send_reply("503", "Service Unavailable");
> exit;
> }
>
>
> #1 - tests if client has a private IP address (as defined by RFC1918)
> #in the Contact field of the SIP message.
> #2 - tests if client has contacted OpenSIPS from an address that is
> different
> #from the one in the Via field. Both the IP and port are compared by this
> test.
> #4 - tests if client has a private IP address (as defined by RFC1918) in
> the top
> #Via field of the SIP message.
>
>
> if (client_nat_test("3")) {
> fix_contact();
>
> if ((method=="REGISTER" ||(method=="INVITE" && !has_totag())) )
> {
> nat_keepalive();
> }
> }
>
> # check if user is suspended
> if(is_method("REGISTER|INVITE|OPTIONS"))
> {
> if (radius_is_user_in("From", "suspended")) {
> sl_send_reply("403", "Forbidden - suspended");
> exit;
> };
> };
>
> #use engate media proxy to fully control the media
> if (method==INVITE && (client_nat_test("3") ||
> search("^Route:.*;nat=yes")) ) {
> engage_media_proxy();
> }
>
> #has_totag() indicate in-dialog request. all in dialog request are
> processed in this block
> if (has_totag()) {
> # sequential request withing a dialog should
> # take the path determined by record-routing
>
> #loose_route() is used to route is usually used to
> #route in-dialog requests (like ACK, BYE, reINVITE).
> #The loose_route function analyzes the Route: headers in the requests.
> #If there is no Route: header, the function returns FALSE and routing
> #should be done with normal lookup functions. If a Route: header is found,
> #the function returns 1 and behaves as described in section 16.12 of RFC
> 3261.
> #There is only one exception: If the request is out-of-dialog (no to-tag)
> #and there is only one Route: header indicating the local proxy,
> #then the Route: header is removed and the function returns FALSE.
> if (loose_route()) {
> # mark routing logic in request
> append_hf("P-hint: rr-enforced\r\n");
>
> #some provider GW (incorrectly) updated the contact info of an
> established dialog when it got an ACK. fix it
> if(is_method("ACK")) {
> if(is_present_hf("Contact")) remove_hf("Contact");
> };
> route(1);
> } else {
>
> if ( is_method("ACK") ) {
> #t_check_trans Returns true if the current request is associated to a
> transaction
> 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.\n");
> xlog("L_WARN", "[$mi] discarding ACK\n");
> exit;
> }
> }
> #in-dialog , not loose route, and not ACK, we discard.
> sl_send_reply("404","Not here");
> }
> #regardless of whatever happens, all in-dialog has to end here.
> exit;
> }
>
> t_check_trans();
>
> # CANCEL processing.
> if (is_method("CANCEL"))
> {
> setflag(1); # do accounting ...
> setflag(2); # sip trace
> #Returns true if the current request is associated to a transaction.
> #CANCEL request - true if the cancelled INVITE transaction exists
> if (t_check_trans())
> t_relay();
> exit;
> }
>
> #following must be initial requests, or register.
> #command must be INVITE, ACK, BYE, OPTIONS, REGISTER
>
> # authenticate if from local subscriber (uncomment to enable auth)
> #if (!(method=="REGISTER") && from_uri==myself)
> #{
> # if (!proxy_authorize("", "subscriber")) {
> # proxy_challenge("", "0");
> # exit;
> # }
> # if (!check_from()) {
> # sl_send_reply("403","Forbidden auth ID");
> # exit;
> # }
>
> # consume_credentials();
> # # caller authenticated
> #}
>
>
> # record routing
> if (!is_method("REGISTER"))
> record_route();
>
> # account only INVITEs
> if (is_method("INVITE")) {
> setflag(1); # do accounting
> setflag(2); # sip trace
> }
>
> #fraud detection block. we don't allow outsiders who are not authenticated
> to use our gateway.
> if (!uri==myself)
> /* replace with following line if multi-domain support is used */
> ##if (!is_uri_host_local())
> {
> # check if user is allowed to do voip calls to other domains
> if(is_method("INVITE")) {
> #for caller calling outside, but not in our voip group, we
> forbid.
> #this is needed to fight against fraud.
> if (!radius_is_user_in("From", "voip")) {
> sl_send_reply("403", "Forbidden VoIP");
> exit;
> };
> };
> # mark routing logic in request
> append_hf("P-hint: outbound\r\n");
>
> route(1);
> exit;
> }
>
>
> #process REGISTER to local server.
> if (is_method("REGISTER"))
> {
> # authenticate the REGISTER requests (uncomment to enable auth)
> if (!radius_www_authorize("machinename.somedomain.com")
> && !www_authorize("machinename.somedomain.com", "subscriber"))
> {
> www_challenge("machinename.somedomain.com", "0");
> exit;
> }
> ##
> ##if (!check_to())
> ##{
> ## sl_send_reply("403","Forbidden auth ID");
> ## exit;
> ##}
>
> if (client_nat_test("3")) fix_nated_register();
> if (!save("location"))
> sl_reply_error();
> exit;
> }
>
> #process INVITE, ACK, BYE, OPTIONS for local server in the following
> blocks
>
> if ($rU==NULL) {
> # request with no Username in RURI
> sl_send_reply("484","Address Incomplete");
> exit;
> }
>
>
> #lookup(domain) extracts username from Request-URI and tries to find
> #all contacts for the username in usrloc
> #return codes
> #1 - contacts found and returned.
> #-1 - no contact found.
> #-2 - contacts found, but method not supported.
> #-3 - internal error during processing
> #if (!lookup("location")) {
> # switch ($retcode) {
> # case -1:
> # case -3:
> # t_newtran();
> # t_reply("404", "Not Found");
> # exit;
> # case -2:
> # sl_send_reply("405", "Method Not Allowed");
> # exit;
> # }
> #}
>
>
> #process INVITE, ACK, BYE, OPTIONS for local server in the following
> blocks
> #It is critical to save $avp(s:can_uri) after the Proxy has performed
> #all possible lookups except DNS.
> #The Canonical-URI will be used for rating the session.
> $avp(s:can_uri) = $ru;
> route(1);
> }
>
>
> #route[1] process INVITE, ACK, BYE, OPTIONS
> route[1] {
>
>
> if (is_method("INVITE") ) {
>
> # normalization to e164
> # http://en.wikipedia.org/wiki/NANP
> if($ruri.user =~ "^\+[1-9][0-9]+")
> {
> strip(1);
> }
> # if($ruri.user =~ "^00[1-9][0-9]+") {
> # strip(2);
> # }
> # if($ruri.user =~ "^0[1-9][0-9]+") {
> # strip(1);
> # #prefix("49");
> # }
>
>
> #in the US , dialing 1NPANXXXXXX 11 digits
> if($ruri.user =~ "^1[1-9][0-9]{9}") {
> #do nothing
> }
> #in the US, dialing NPANXXXXXX 10 digits
> else if($ruri.user =~ "^[1-9][0-9]{9}") {
> prefix("1");
> }
> #in the US, dialing NXXXXXX 7 digits local number.
> else if($ruri.user =~ "^[1-9][0-9]{6}") {
> $rU = $(fU{s.substr, 0, 4}) + $rU;
> }
> # 411 Local Directory Assistance
> else if (uri=~"^sip:411 at .*") {
> # the uri with a default call to "local directory
> assistance".
> $rU = $(fU{s.substr, 0, 4}) + "5551212";
> }
>
> # 611 Local Directory Assistance
> else if (uri=~"^sip:611 at .*") {
> # the uri with a default call to "local directory
> assistance".
> rewriteuri("sip:17775551212 at machinename.somedomain.com");
> }
> #911 is handled by E911 service provider
> else {
> sl_send_reply("404", "Invalid destination");
> exit;
> }
>
> # Set the callerid for the user from an AVP
> #if (avp_db_load("$from/username", "s:callerid")) {
> # subst('/^From: (.*)>(.*)$/From: $avp(callerid)>\2/ig');
> #};
>
> }
>
> if (is_method("INVITE|BYE")) {
> setflag(1); # do accounting ...
> setflag(2); # sip trace
> #call the accounting functions explicitly in local_route for
> #the internally generated BYEs as they do not trigger accounting by just
> #setting the accounting flag
> acc_rad_request("200 ok");
> acc_log_request("200 ok");
> }
>
> #change access point phone number to inbound route for asterisk
> alias_db_lookup("dbaliases");
> #forward asterisk inbound route with dispatcher as load balancer
> if (is_method("INVITE") && $ruri =~ "^sip:17771000101 at .*" ) {
> #dispatcher select from set 1 using algorithm 0.
> if(!ds_select_dst("1", "0"))
> {
> sl_send_reply("404", "no destination");
> }
> if(!t_relay()) sl_reply_error();
> exit;
> };
>
> #special relaying to asterisk finished, now we process regular
> requests.
>
> #INVITE, ACK, BYE, OPTIONS to locally registered user.
> if (lookup("location"))
> {
> if (is_method("INVITE")) {
> t_on_branch("1");
> t_on_reply("1");
> t_on_failure("1");
> }
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> }
>
> #INVITE to outgoing gateway, route it out.
> if (is_method("INVITE") ) {
> #if (cr_route("default", "machinename.somedomain.com",
> "$rU", "$rU", "call_id")) {
> if (do_routing()) {
> t_on_failure("11");
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> };
> exit;
> };
>
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> }
>
>
> branch_route[1] {
> xlog("new branch at $ru\n");
> }
>
>
> onreply_route[1] {
> xlog("incoming reply\n");
> }
>
>
> failure_route[1] {
> if (t_was_cancelled()) {
> exit;
> }
>
> # uncomment the following lines if you want to block client
> # redirect based on 3xx replies.
> ##if (t_check_status("3[0-9][0-9]")) {
> ##t_reply("404","Not found");
> ## exit;
> ##}
>
> # uncomment the following lines if you want to redirect the failed
> # calls to a different new destination
> ##if (t_check_status("486|408")) {
> ## sethostport("192.168.2.100:5060");
> ## append_branch();
> ## # do not set the missed call flag again
> ## t_relay();
> ##}
> }
>
>
> ######################
> # "default" failover #
> ######################
> failure_route[11] {
> xlog("L_INFO", "entering failure_route[11] for reply code
> '$T_reply_code'\n");
>
> if (t_was_cancelled()) {
> exit;
> }
>
> if (t_check_status("408|5[0-9][0-9]")) {
> #xlog("L_INFO","cr_tree_rewrite_uri(\"default\", \"1\");\n");
> #if (cr_route("default", "machinename.somedomain.com", "$rU", "$rU",
> "call_id")) {
> if (do_routing()) {
> t_on_failure("12");
> append_branch();
> route(1);
> };
> exit;
> } else if (t_check_status("3[0-9][0-9]")) {
> t_reply("404","Not found");
> exit;
> }
> }
>
> failure_route[12] {
> xlog("L_INFO", "entering failure_route[12] for reply code
> '$T_reply_code'\n");
> if (t_was_cancelled()) {
> exit;
> }
>
> if (t_check_status("408|5[0-9][0-9]")) {
> xlog("L_INFO","cr_tree_rewrite_uri(\"default\", \"2\");\n");
> #if (cr_route("default", "machinename.somedomain.com", "$rU", "$rU",
> "call_id")) {
> if (do_routing()) {
> t_on_failure("13");
> append_branch();
> route(1);
> };
> exit;
> } else if (t_check_status("3[0-9][0-9]")) {
> t_reply("404","Not found");
> exit;
> }
> }
>
> failure_route[13] {
> xlog("L_INFO", "entering failure_route[13] for reply code
> '$T_reply_code'\n");
> if (t_was_cancelled()) {
> exit;
> }
> }
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



May be if i run into problems, could i also seek your help????


-- 
Khan


VoIP Rookie
Every beginning has an end regardless we believe it or not...