[OpenSIPS-Users] Change in SDP/RTP routing with Opensips 1.4 and 1.6
Ovidiu Sas
osas at voipembedded.com
Mon Feb 7 19:22:03 CET 2011
Just capture a call that is going through your SIP proxy and check the
SDP in the received and sent INVITE and the SDP in the received and
sent 200ok. The connection IP and port for each SDP should be the
same (untouched).
If it's untouched, then your opensips config is working as expected
and your RTP problem is somewhere else.
Regards,
Ovidiu Sas
On Mon, Feb 7, 2011 at 1:10 PM, Chris Stone <axisml at gmail.com> wrote:
> On Mon, Feb 7, 2011 at 10:35 AM, Ovidiu Sas <osas at voipembedded.com> wrote:
>> You are re-posting the same question again without providing any
>> additional info:
>> http://lists.opensips.org/pipermail/users/2011-February/016626.html
>
> Sorry, did not see my original message in my Gmail sent messages, or a
> reply - so, figured it did not get sent and resent.
>
>> Have you checked if the your config is altering the SDP?
>> Check the SDP for the same message before coming to your opensips
>> server and after leaving the opensips server.
>> If the SDP is altered and the IP in SDP is pointing to your opensips
>> server, then there is your problem.
>
> The config does nothing with the SDP. The config is nearly identical
> (is identical functionality wise) to what we're running with Opensips
> 1.4. Was there some 'default' behavior change between 1.4 and 1.6 that
> might be the cause?
>
> For reference, here's our config for the 1.6 installation:
>
> ####### Global Parameters #########
>
> #debug=3
> debug=9
> log_stderror=no
> #log_facility=LOG_LOCAL0
>
> fork=yes
> children=25
>
> /* 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
>
> sip_warning=1
> tos=IPTOS_LOWDELAY
>
> ####### Modules Section ########
>
> #set module path
> mpath="/usr/lib64/opensips/modules"
>
> /* uncomment next line for MySQL DB support */
> loadmodule "signaling.so"
> loadmodule "sl.so"
> loadmodule "tm.so"
> loadmodule "rr.so"
> loadmodule "maxfwd.so"
> loadmodule "textops.so"
> loadmodule "mi_fifo.so"
> loadmodule "uri.so"
> loadmodule "acc.so"
> loadmodule "avpops.so"
>
> loadmodule "perl.so"
>
> # originate/terminate permissions
> loadmodule "permissions.so"
>
> # dispatcher
> loadmodule "dispatcher.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)
> # do not append from tag to the RR (no need for this script)
> modparam("rr", "append_fromtag", 0)
>
>
> # ----- uri params -----
> modparam("uri", "use_uri_table", 0)
>
>
> # ----- 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)
> /* account triggers (flags) */
> modparam("acc", "failed_transaction_flag", 3)
> modparam("acc", "log_flag", 1)
> modparam("acc", "log_missed_flag", 2)
> /* uncomment the following lines to enable DB accounting also */
> modparam("acc", "db_flag", 1)
> modparam("acc", "db_missed_flag", 2)
>
>
> # ----- perl params -----
> modparam("perl", "modpath", "/usr/lib64/opensips/modules/")
> modparam("perl", "filename", "/usr/local/sbin/perl_sql_routing.pl")
>
>
> # ----- permissions params -----
> modparam("permissions", "default_deny_file", "/etc/opensips/permissions.deny")
> modparam("permissions", "default_allow_file", "/etc/opensips/permissions.allow")
>
>
> # ----- dispatcher params ----
> modparam("dispatcher","list_file","/etc/opensips/dispatcher.list")
> modparam("dispatcher","force_dst",1)
> modparam("dispatcher","flags",3)
> modparam("dispatcher", "dst_avp", "$avp(i:271)")
> modparam("dispatcher", "grp_avp", "$avp(i:272)")
> modparam("dispatcher", "cnt_avp", "$avp(i:273)")
>
>
> ####### Routing Logic ########
>
>
> # main request routing logic
>
> route{
>
> if (!mf_process_maxfwd_header("32")) {
> xlog("L_ERR", " <$rm> from <$fu> to <$tu> r-uri <$ru>
> src_ip <$si> [483 - Too Many Hops]\n");
> sl_send_reply("483","Too Many Hops");
> exit;
> }
>
> if (msg:len > 16384 ) {
> xlog("L_ERR", " <$rm> from <$fu> to <$tu> r-uri <$ru>
> src_ip <$si> [413 - Message is too big - greater than 16384
> bytes]\n");
> sl_send_reply("413", "Message is too big - greater
> than 16384 bytes]");
> exit;
> };
>
> if (is_method("REGISTER")) {
> xlog("L_ERR", " <$rm> from <$fu> to <$tu> r-uri <$ru>
> src_ip <$si> [503 - Registration Unavailable]\n");
> sl_send_reply("503"," Registration Unavailable via
> this proxy - contact support at axint.net for assistance");
> exit;
> }
>
> if (is_method("PUBLISH"))
> {
> xlog("L_ERR", " <$rm> from <$fu> to <$tu> r-uri <$ru>
> src_ip <$si> [503 - Publish Service Unavailable]\n");
> sl_send_reply("503", "Service Unavailable");
> exit;
> }
>
> if (has_totag()) {
> #xlog("L_ERR", "Request has_totag(). Request: <$rm>
> from <$fu> to <$tu> r-uri <$ru> src_ip <$si>\n");
> # sequential request withing a dialog should
> # take the path determined by record-routing
> if (loose_route()) {
> if (is_method("BYE")) {
> #xlog("L_ERR", "Request is BYE - doing
> accounting. Request: <$rm> from <$fu> to <$tu> r-uri <$ru> src_ip
> <$si>\n");
> setflag(1); # do accounting ...
> setflag(3); # ... even if the transaction fails
> } else if (is_method("INVITE")) {
> #xlog("L_ERR", "Request is INVITE -
> recording route. Request: <$rm> from <$fu> to <$tu> r-uri <$ru> src_ip
> <$si>\n");
> # 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 $du (destination URI).
> #xlog("L_ERR", " <$rm> from <$fu> to <$tu>
> r-uri <$ru> src_ip <$si> route(1 - BYE or reINVITE)\n");
> route(1);
> } else {
> /* uncomment the following lines if you want
> to enable presence */
> ##if (is_method("SUBSCRIBE") && $rd ==
> "your.server.ip.address") {
> ## # in-dialog subscribe requests
> ## route(2);
> ## exit;
> ##}
> if ( is_method("ACK") ) {
> if ( t_check_trans() ) {
> #xlog("L_ERR", " <$rm> from
> <$fu> to <$tu> r-uri <$ru> src_ip <$si> route(relay ACK)\n");
> # non loose-route, but
> stateful ACK; must be an ACK after
> # a 487 or e.g. 404 from upstream server
> t_relay();
> exit;
> } else {
> #xlog("L_ERR", " <$rm> from
> <$fu> to <$tu> r-uri <$ru> src_ip <$si> ruote(none - discarding
> unmatched ACK)\n");
> # ACK without matching transaction ->
> # ignore and discard
> exit;
> }
> }
> xlog("L_ERR", " <$rm> from <$fu> to <$tu>
> r-uri <$ru> src_ip <$si> [404 - Not here]\n");
> sl_send_reply("404","Not here");
> }
> exit;
> }
>
> #initial requests
>
> # CANCEL processing
> if (is_method("CANCEL"))
> {
> xlog("L_ERR", " <$rm> from <$fu> to <$tu> r-uri <$ru>
> src_ip <$si> [200 - CANCEL request processing]\n");
> if (t_check_trans())
> t_relay();
> exit;
> }
>
> t_check_trans();
>
> # preloaded route checking
> if (loose_route()) {
> #xlog("L_ERR", "loose_route(), so add P_hint
> rr-enforced header and route request: <$rm> from <$fu> to <$tu> r-uri
> <$ru> src_ip <$si>\n");
> # mark routing logic in request
> append_hf("P-hint: rr-enforced\r\n");
> route(1);
> exit;
> }
>
> # record routing
> if (!is_method("REGISTER|MESSAGE")) {
> #xlog("L_ERR", "Not REGISTER or MESSAGE, so recording
> route for request: <$rm> from <$fu> to <$tu> r-uri <$ru> src_ip
> <$si>\n");
> record_route();
> }
>
> # account only INVITEs
> if (is_method("INVITE")) {
> #xlog("L_ERR", "Is INVITE so do accounting for
> request: <$rm> from <$fu> to <$tu> r-uri <$ru> src_ip <$si>\n");
> setflag(1); # do accounting
> }
>
> # do lookup with method filtering
> # if (!lookup("location","m")) {
> # switch ($retcode) {
> # case -1:
> # case -3:
> # t_newtran();
> # t_reply("404", "Not Found");
> # exit;
> # case -2:
> # sl_send_reply("405", "Method Not Allowed");
> # exit;
> # }
> # }
>
> if (is_method("ACK"))
> {
> xlog("L_ERR", "ACK - <$rm> from <$fu> to <$tu> r-uri
> <$ru> src_ip <$si> [200 - OK]\n");
> if (t_check_trans())
> t_relay();
> exit;
> }
>
> # verify user has rights to originate/terminate calls
> if (is_method("INVITE")) {
> if (!allow_routing()) {
> xlog("L_ERR", " <$rm> from <$fu> to <$tu>
> r-uri <$ru> src_ip <$si> [503 Service Unavailable (route denied)]\n");
> sl_send_reply("503", " Service unavailable at
> this time");
> exit;
> };
> xlog("L_ERR", " ORIGINAL-INVITE [method=$rm, from=$fu,
> to=$ru, src_ip=$si]\n");
> };
>
> # when routing via usrloc, log the missed calls also
> setflag(2);
>
> # prefix 1-areacode if seven digit dialing - use areacode from
> the from_uri (caller)
> if (uri=~"^sip:[0-9]{7}@") {
> if (from_uri=~"^sip:1303[0-9]{7}@") {
> prefix("+1303");
> } else if (from_uri=~"^sip:1720[0-9]{7}@") {
> prefix("+1720");
> } else if (from_uri=~"^sip:1970[0-9]{7}@") {
> prefix("+1970");
> } else if (from_uri=~"^sip:1925[0-9]{7}@") {
> prefix("+1925");
> } else if (from_uri=~"^sip:1312[0-9]{7}@") {
> prefix("+1312");
> } else if (from_uri=~"^sip:1406[0-9]{7}@") {
> prefix("+1406");
> } else if (from_uri=~"^sip:1415[0-9]{7}@") {
> prefix("+1415");
> } else if (from_uri=~"^sip:1602[0-9]{7}@") {
> prefix("+1602");
> } else if (from_uri=~"^sip:1631[0-9]{7}@") {
> prefix("+1631");
> } else if (from_uri=~"^sip:1617[0-9]{7}@") {
> prefix("+1617");
> } else if (from_uri=~"^sip:1909[0-9]{7}@") {
> prefix("+1909");
> } else if (from_uri=~"^sip:1925[0-9]{7}@") {
> prefix("+1925");
> } else if (from_uri=~"^sip:1305[0-9]{7}@") {
> prefix("+1305");
> } else if (from_uri=~"^sip:1768[0-9]{7}@") {
> prefix("+1768");
> } else if (from_uri=~"^sip:1505[0-9]{7}@") {
> prefix("+1505");
> } else if (from_uri=~"^sip:1254[0-9]{7}@") {
> prefix("+1254");
> } else if (from_uri=~"^sip:1480[0-9]{7}@") {
> prefix("+1480");
> };
> };
>
> # prefix 1 for 10 digits dialed
> if (uri =~ "^sip:[0-9]{10}@") {
> prefix("+1");
> };
>
> # make sure the destination number has a + prefix
> if (uri =~ "^sip:[0-9]{7,}@") {
> prefix("+");
> };
>
> # determine call route to take from sql query in perl addin
> module script
> $avp(i:369) = 10;
> perl_exec("get_route_sql");
> # avp_print();
> if ($avp(i:369) == "1") {
> route(1); # route transaction replies
> exit;
> } else if ($avp(i:369) == "2") {
> route(2); # route to voip01
> exit;
> } else if ($avp(i:369) == "3") {
> route(3); # route to voip02
> exit;
> } else if ($avp(i:369) == "4") {
> route(4); # route reply error
> exit;
> } else if ($avp(i:369) == "5") {
> # 2009-11-29 (cstone) - if 7, 10 or 11 digits on
> destination number - route out to DASH
> if (uri =~ "^sip:\+?[0-9]{7}@") {
> xlog("L_ERR", "INVITE - 7 DIGIT DESTINATION - FORCED
> route to DASH\n");
> route(10);
> exit;
> }
> if (uri =~ "^sip:\+?[0-9]{10,11}@") {
> xlog("L_ERR", "INVITE - 10 or 11 DIGIT DESTINATION -
> FORCED route to DASH\n");
> route(10);
> exit;
> }
> if (uri =~ "^sip:\+?411@") {
> xlog("L_ERR", "INVITE - 411 DESTINATION - FORCED route
> to DASH\n");
> route(10);
> exit;
> }
> # continue on through verizon
> route(5); # route out to Verizon
> exit;
> } else if ($avp(i:369) == "6") {
> route(6); # route out to GlobalPOP
> exit;
> } else if ($avp(i:369) == "7") {
> route(7); # route to tranquility networks server 1
> exit;
> } else if ($avp(i:369) == "8") {
> route(8); # route to inmate-relay server
> exit;
> } else if ($avp(i:369) == "9") {
> route(9); # route to west-fax server
> exit;
> } else if ($avp(i:369) == "10") {
> route(10); # route to out to dash
> exit;
> } else if ($avp(i:369) == "11") {
> route(11); # route to demo server
> exit;
> } else if ($avp(i:369) == "12") {
> route(12); # route to CoroWare server
> exit;
> } else if ($avp(i:369) == "13") {
> route(13); # CALL KILLER ROUTE
> exit;
> } else if ($avp(i:369) == "14") {
> route(14); # route to old inmate-relay server
> exit;
> } else if ($avp(i:369) == "15") {
> route(15); # route to voip02 server
> exit;
> } else if ($avp(i:369) == "25") {
> route(25); # route to new venus
> exit;
> }
>
> if (is_method("INVITE|BYE")) {
> xlog("L_ERR", "USING DEFAULT FALL THROUGH ROUTING TO DASH\n");
> route(10);
> exit;
> };
>
> route(1);
> }
>
>
> route[1] {
> # for INVITEs enable some additional helper routes
> if (is_method("INVITE")) {
> t_on_branch("2");
> t_on_reply("2");
> t_on_failure("1");
> }
>
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> }
>
>
> route[2] {
> # strip the leading + sign from the To: uri
> subst_uri('/^sip:\+([0-9]+)@(.*)$/sip:\1@\2/i');
> replace_all("sip:\+","sip:");
>
> # send call to server voip01.axint.net
> xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(in to voip01)\n");
> forward("10.10.10.10");
> exit;
> }
>
> route[3] {
> # strip the leading + sign from the To: uri
> subst_uri('/^sip:\+([0-9]+)@(.*)$/sip:\1@\2/i');
> replace_all("sip:\+","sip:");
>
> # send call to server voip02.axint.net
> if (is_method("INVITE|BYE|CANCEL")) {
> xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(in to voip02)\n");
> };
> forward("10.10.10.11");
> exit;
> }
>
> route[4] {
> #if (!save("location")) {
> sl_reply_error();
> #};
> }
>
> # send calls to MCI/Verizon for routing to PSTN network (or other VoIP networks)
> route[5] {
> if (is_method("OPTIONS")) {
> xlog("L_ERR", " <$rm> from <$fu> to <$tu> r-uri <$ru>
> src_ip <$si> [503 OPTIONS to gateway invalid - not forwarding to
> verizon]");
> sl_send_reply("503"," SIP OPTIONS packet to gateway
> invalid - contact support at axint.net for assistance");
> exit;
> }
>
> # Add P-Asserted-Identity header field if not already present
> if (!is_present_hf("P-Asserted-Identity")) {
> xlog("L_ERR", " MCI/Verizon route - P-Asserted-Identity header
> missing - adding to SIP request\n");
> $avp(i:678) = $fu;
> avp_subst("$avp(i:678)", "/sip:([0-9]+)@(.*)>?$/<sip:+\1@\2:5060>/");
> #avp_print();
> append_hf("P-Asserted-Identity: $avp(i:678)\r\n");
> };
>
> ds_select_domain("1","4");
>
> # log the call information
> xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(out to verizon)\n");
>
> forward();
> exit;
> }
>
> # send calls to GlobalPOP server (TESTING-TESTING)
> route[6] {
> # log the call information
> xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(out to GlobalPOP)\n");
>
> forward("10.0.0.10");
> exit;
> }
>
> route[7] {
> # strip the leading + sign from the To: uri
> subst_uri('/^sip:\+([0-9]+)@(.*)$/sip:\1@\2/i');
> replace_all("sip:\+","sip:");
>
> # send call to server Tranquility Networks Asterisk server
> if (is_method("INVITE|BYE|CANCEL")) {
> xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(in to tranquility)\n");
> };
> forward("10.10.0.10");
> exit;
> }
>
> # send calls to inmate-relay server
> route[8] {
> # strip the leading + sign from the To: uri
> subst_uri('/^sip:\+([0-9]+)@(.*)$/sip:\1@\2/i');
> replace_all("sip:\+","sip:");
>
> # send call to server inmate-relay.axint.net
> #if (is_method("INVITE|BYE|CANCEL")) {
> xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(in to inmate-relay)\n");
> #};
> forward("10.10.10.12");
> exit;
> }
>
> # send calls to West FAX server
> route[9] {
> # strip the leading + sign from the To: uri
> subst_uri('/^sip:\+([0-9]+)@(.*)$/sip:\1@\2/i');
> replace_all("sip:\+","sip:");
>
> # send call to server West Fax Asterisk server
> if (is_method("INVITE|BYE|CANCEL")) {
> xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(in to west-fax)\n");
> };
> forward("10.10.100.10");
> exit;
> }
>
> # send calls to Dash for routing to PSTN network (or other VoIP networks)
> route[10] {
> if (is_method("OPTIONS")) {
> xlog("L_ERR", " <$rm> from <$fu> to <$tu> r-uri <$ru>
> src_ip <$si> [503 OPTIONS to gateway invalid - not forwarding to
> dash]");
> sl_send_reply("503"," SIP OPTIONS packet to gateway
> invalid - contact support at axint.net for assistance");
> exit;
> }
>
> #xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(preparing to route to dash)\n");
>
> ds_select_domain("2","4");
>
> # log the call information
> xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(out to dash)\n");
>
> forward();
> exit;
> }
>
> # send calls to Demo server
> route[11] {
> # strip the leading + sign from the To: uri
> subst_uri('/^sip:\+([0-9]+)@(.*)$/sip:\1@\2/i');
> replace_all("sip:\+","sip:");
>
> # send call to server ScanOne server
> if (is_method("INVITE|BYE|CANCEL")) {
> xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(in to demo server)\n");
> };
> forward("10.10.10.13");
> exit;
> }
>
> # send calls to CoroWare server
> route[12] {
> # strip the leading + sign from the To: uri
> subst_uri('/^sip:\+([0-9]+)@(.*)$/sip:\1@\2/i');
> replace_all("sip:\+","sip:");
>
> # send call to server ScanOne server
> if (is_method("INVITE|BYE|CANCEL")) {
> xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(in to CoroWare)\n");
> };
> forward("10.100.100.10");
> exit;
> }
>
> # CALL KILLER ROUTE
> route[13] {
> xlog("L_ERR", " <$rm> from <$fu> to <$tu> r-uri <$ru> src_ip
> <$si> [604 Does not exist anywhere - prohibited destination]\n");
> sl_send_reply("604","Does not exist anywhere - prohibited destination");
> exit;
> }
>
> # send calls to inmate-relay server
> route[14] {
> # strip the leading + sign from the To: uri
> subst_uri('/^sip:\+([0-9]+)@(.*)$/sip:\1@\2/i');
> replace_all("sip:\+","sip:");
>
> # send call to server inmate-relay.axint.net
> if (is_method("INVITE|BYE|CANCEL")) {
> xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(in to old-inmate-relay)\n");
> };
> forward("10.10.10.15");
> exit;
> }
>
> # send calls to voip02 server
> route[15] {
> # strip the leading + sign from the To: uri
> subst_uri('/^sip:\+([0-9]+)@(.*)$/sip:\1@\2/i');
> replace_all("sip:\+","sip:");
>
> # send call to server
> if (is_method("INVITE|BYE|CANCEL")) {
> xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(in to new-voip02)\n");
> };
> forward("10.10.10.20");
> exit;
> }
>
> # send call to server at PP
> route[25] {
> # strip the leading + sign from the To: uri
> subst_uri('/^sip:\+([0-9]+)@(.*)$/sip:\1@\2/i');
> replace_all("sip:\+","sip:");
>
> # send call to server
> if (is_method("INVITE|BYE|CANCEL")) {
> xlog("L_ERR", " [method=$rm, from=$fu, to=$ru, src_ip=$si] =>
> route(in to venus)\n");
> };
>
> if (is_method("INVITE")) {
> create_dialog();
> engage_media_proxy();
> }
>
> forward("10.10.10.25");
> exit;
> }
>
>
> branch_route[2] {
> xlog("new branch at $ru\n");
> }
>
>
> onreply_route[2] {
> xlog("incoming reply\n");
> }
>
>
> failure_route[1] {
> if (t_was_cancelled()) {
> exit;
> }
>
> # uncomment the following lines if you want to block client
> # redirect based on 3xx replies.
> ##if (t_check_status("3[0-9][0-9]")) {
> ##t_reply("404","Not found");
> ## exit;
> ##}
>
> # uncomment the following lines if you want to redirect the failed
> # calls to a different new destination
> ##if (t_check_status("486|408")) {
> ## sethostport("192.168.2.100:5060");
> ## # do not set the missed call flag again
> ## t_relay();
> ##}
> }
>
> error_route {
> xlog("--- error route class=$(err.class) level=$(err.level)
> info=$(err.info) rcode=$(err.rcode) rreason=$(err.rreason) ---\n");
> xlog("--- error from [$si:$sp]\n+++++\n$mb\n++++\n");
> sl_send_reply("$err.rcode", "$err.rreason");
> exit;
> }
>
>
>
>
> Chris
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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