[OpenSIPS-Users] OCS-Asterisk Gateway tcp-to-udp
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Tue Nov 11 11:32:25 CET 2008
Hi Steffen,
Before sending the INVITE to OCS, do on your opensips a
sethostport(OCS_IP); - this will replace the address of proxy (from
RURI) with the one of the OCS server..
Regards,
Bogdan
Steffen Görlach wrote:
> Hello Newsgroup,
>
> I still got some issues with my OCS server.
> Ips:
> x.x.116.2 Mediation Port:5060
> x.x.116.15 Asterisk P:5060
> x.x.116.16 Opensips P:5061
>
> If I send from my Asterisk the SIP/UDP message destinated to Opensips IP, the opensips relays this packet to SIP/TCP to the OCS-Mediation Server,
> So long so good.
> But the SIP Information is still addressed to the Opensips, so the OCS won't accept this SIP-Message.
> Log:
> 192.168.116.16 -> 192.168.116.2 SIP/SDP Request: INVITE sip:192.168.116.16:5061, with session description
> Error:
> 192.168.116.2 -> 192.168.116.16 SIP Status: 400 Get Request/To Sip Uri error.
>
> Is there a way, to tell opensips, he should also change the SIP-Addresses to the point he relays it?
> .
>
> My actual opensips.cfg:
>
> # main request routing logic
>
> route{
>
> xlog( "L_ERR", "LOG: ROU time=[$Tf] contact header=[$ct] from uri=[$fu] request's method=[$rm] request's uri=[$ru] to uri=[$tu] sip message id=[$mi] ip source=[$src_ip]\n" );
>
> if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","Too Many Hops");
> exit;
> }
>
> if (src_ip == 192.168.116.2) {
> #remove misleading CONTACT header line
> remove_hf("Contact");
> #remove UTF-8 information, as * is not able to process it properly
> subst("/^(CONTENT-TYPE:.*);[ ]*charset=utf-8(.*)/\1\2/");
> #relay request to *
> if (!t_relay("udp:192.168.116.15:5060")) {
> xlog("LOG: Goto asterisk \n");
> sl_reply_error();
> }
> xlog( "L_ERR", "LOG: AST time=[$Tf] contact header=[$ct] from uri=[$fu] request's method=[$rm] request's uri=[$ru] to uri=[$tu] sip message id=[$mi] ip source=[$src_ip]\n" );
> }
> else {
> #relay request to OCS
> if (!t_relay("tcp:192.168.116.2:5060") ) {
> xlog("LOG: Return to OCS \n");
> sl_reply_error();
> }
> xlog( "L_ERR", "LOG: OCS time=[$Tf] contact header=[$ct] from uri=[$fu] request's method=[$rm] request's uri=[$ru] to uri=[$tu] sip message id=[$mi] ip source=[$src_ip]\n" );
> }
>
> }
>
> onreply_route {
>
> xlog( "L_ERR", "LOG: Orr time=[$Tf] contact header=[$ct] from uri=[$fu] request's method=[$rm] request's uri=[$ru] to uri=[$tu] sip message id=[$mi] ip source=[$src_ip]\n" );
>
> exit;
> }
>
> branch_route[1] {
> xlog("LOG:new branch at $ru\n");
> }
>
> failure_route[1] {
> if (t_was_cancelled()) {
> exit;
> }
> }
> -----Ursprüngliche Nachricht-----
> Von: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] Im Auftrag von Steffen Görlach
> Gesendet: Freitag, 24. Oktober 2008 15:57
> An: Bogdan-Andrei Iancu
> Cc: users at lists.opensips.org
> Betreff: Re: [OpenSIPS-Users] OCS-Asterisk Gateway tcp-to-udp
>
> Hello Bogdan,
>
> thanks for your quick response.
> I commented out the route[1] and now, everything works.
> Thx for your help.
>
> Regards,
> Steffen
>
> -----Ursprüngliche Nachricht-----
> Von: Bogdan-Andrei Iancu [mailto:bogdan at voice-system.ro]
> Gesendet: Freitag, 24. Oktober 2008 14:34
> An: Steffen Görlach
> Cc: users at lists.opensips.org
> Betreff: Re: [OpenSIPS-Users] OCS-Asterisk Gateway tcp-to-udp
>
> Hi Steffen,
>
> Do you get the CANCEL on your opensips ? if so, do you see the xlog
> messages for the CANCEL?
>
> Also looking at your script, I see a duplicate block - you do the
> routing in the main route and again in route[1] - you should remove the
> route[1] from your script.
>
> Regards,
> Bogdan
>
> Steffen Görlach wrote:
>
>> Hello to all,
>>
>> I`m trying to setup opensips as a gateway from Office Communication
>> Server (tcp) to Asterisk (udp). I followed this guide:
>> http://confluence.terena.org:8080/display/IPTelCB/3.2.7.+Tying+MS+OCS+with+Asterisk+through+OpenSER
>>
>> My Setup is:
>>
>> Trixbox with 2 IPs:
>>
>> 192.168.67.4 -> Asterisk (Port 5060)
>>
>> 192.168.67.5 -> Opensips (Port 5060)
>>
>> Office Communication Server 2007
>>
>> 192.168.67.83 -> OCS-Mediation-Server (Port 5060)
>>
>> This Setup can establish phone calls , but I got problems with the
>> SIP-Status messages. If I call from OCS-Phone to Asterisk Extension
>> and I cancel the Call, the Asterisk Phone don’t get this Cancel-Message.
>>
>> *How can I route all of the SIP-messages to the Asterisk and the other
>> way?*
>>
>> * *
>>
>> My log posts this ERROR (I don`t if this matters for my Problem):
>>
>> ERROR:core:forward_reply: no 2nd via found in reply
>>
>> * *
>>
>> Here is my opensips.cfg:
>>
>> #
>>
>> # $Id: opensips.cfg 4423 2008-06-27 10:25:01Z henningw $
>>
>> #
>>
>> # 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.
>>
>> #
>>
>> ####### 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=no
>>
>> /* 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
>>
>> /* uncomment the next line to disable the auto discovery of local aliases
>>
>> based on revers DNS on IPs (default on) */
>>
>> auto_aliases=no
>>
>> /* 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=5061
>>
>> /* 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:192.168.67.5:5060
>>
>> listen=tcp:192.168.67.5:5060
>>
>> ####### Modules Section ########
>>
>> #set module path
>>
>> mpath="//lib/opensips/modules/"
>>
>> /* uncomment next line for MySQL DB support */
>>
>> #loadmodule "db_mysql.so"
>>
>> loadmodule "sl.so"
>>
>> loadmodule "tm.so"
>>
>> loadmodule "rr.so"
>>
>> loadmodule "maxfwd.so"
>>
>> loadmodule "usrloc.so"
>>
>> loadmodule "registrar.so"
>>
>> loadmodule "textops.so"
>>
>> loadmodule "mi_fifo.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"
>>
>> # ----------------- 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)
>>
>> # ----- 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)
>>
>> /* 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)
>>
>> ####### Routing Logic ########
>>
>> # main request routing logic
>>
>> route{
>>
>> if (!mf_process_maxfwd_header("10")) {
>>
>> sl_send_reply("483","Too Many Hops");
>>
>> exit;
>>
>> }
>>
>> if (loose_route()) {
>>
>> xlog("LOG: loose_route \n");
>>
>> append_hf("P-hint: rr-enforced\r\n");
>>
>> route(1);
>>
>> };
>>
>> if (!uri==myself)
>>
>> {
>>
>> append_hf("P-hint: outbound\r\n");
>>
>> route(1);
>>
>> }
>>
>> if (src_ip == 192.168.67.83) {
>>
>> #remove misleading CONTACT header line
>>
>> remove_hf("Contact");
>>
>> #remove UTF-8 information, as * is not able to process it properly
>>
>> subst("/^(CONTENT-TYPE:.*);[ ]*charset=utf-8(.*)/\1\2/");
>>
>> #relay request to *
>>
>> if (!t_relay("udp:192.168.67.4:5060")) {
>>
>> xlog("LOG:Goto asterisk \n");
>>
>> sl_reply_error();
>>
>> }
>>
>> }
>>
>> else {
>>
>> #relay request to OCS
>>
>> if (!t_relay("tcp:192.168.67.83:5060") ) {
>>
>> xlog("LOG:Return to OCS \n");
>>
>> sl_reply_error();
>>
>> }
>>
>> }
>>
>> route(1);
>>
>> }
>>
>> route[1] {
>>
>> xlog("LOG: Route 1 $src_ip \n");
>>
>> # If coming from OCS
>>
>> if (src_ip== 192.168.67.83)
>>
>> {
>>
>> if (method=="INVITE|BYE") {
>>
>> xlog("L_INFO", "*** invite from OCS M=$rm RURI=$ru F=$fu T=$tu IP=$si
>> ID=$ci\n");
>>
>> }
>>
>> xlog("LOG: t_relay Asterisk \n $src_ip \n");
>>
>> t_relay("udp:192.168.67.4:5060");
>>
>> }
>>
>> # else coming from Asterisk
>>
>> else {
>>
>> xlog("LOG: t_relay OCS \n $src_ip \n");
>>
>> t_relay("tcp:192.168.67.83:5060");
>>
>> };
>>
>> exit;
>>
>> }
>>
>> onreply_route {
>>
>> xlog("LOG: incoming reply \n $src_ip \n $tu \n");
>>
>> if (src_ip == 192.168.67.83) {
>>
>> #remove misleading CONTACT header line
>>
>> remove_hf("Contact");
>>
>> #remove UTF-8 information, as * is not able to process it properly
>>
>> subst("/^(CONTENT-TYPE:.*);[ ]*charset=utf-8(.*)/\1\2/");
>>
>> }
>>
>> exit;
>>
>> }
>>
>> branch_route[1] {
>>
>> xlog("LOG:new branch at $ru\n");
>>
>> }
>>
>> failure_route[1] {
>>
>> if (t_was_cancelled()) {
>>
>> exit;
>>
>> }
>>
>> }
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
More information about the Users
mailing list