[OpenSIPS-Users] OpenSIPS is not running, Erorr

[Bogdan-Andrei Iancu]

Hi Khan,

your OpenSIPS runs ok - what you see are runtime errors, not startup 
errors.

The errors you see are indicating processing of SIP reply messages that 
could not be routed - they were received with only one VIA and they were 
not matching any local transaction.

Can you identify the SIP replies triggering this error?

Regards,
Bogdan

Khan Friend wrote:
> Hi guys,
>
> I am trying to troubleshoot errors in my OpenSIPS config file but 
> unable to understand what am i doing wrong.
>
> The log file shows as follows:
> Dec 26 21:38:02 [22302] INFO:usrloc:ul_init_locks: locks array size 512
> Dec 26 21:38:02 [22302] INFO:registrar:mod_init: initializing...
> Dec 26 21:38:02 [22302] INFO:textops:mod_init: initializing...
> Dec 26 21:38:02 [22302] INFO:avpops:avpops_init: initializing...
> Dec 26 21:38:02 [22302] INFO:auth:mod_init: initializing...
> Dec 26 21:38:02 [22302] INFO:auth_db:mod_init: initializing...
> Dec 26 21:38:02 [22302] INFO:core:probe_max_receive_buffer: using a 
> UDP receive buffer of 214 kb
> Dec 26 21:38:56 [22303] ERROR:core:forward_reply: no 2nd via found in 
> reply
> Dec 26 21:38:57 [22308] ERROR:core:forward_reply: no 2nd via found in 
> reply
> Dec 26 21:38:58 [22306] ERROR:core:forward_reply: no 2nd via found in 
> reply
> Dec 26 21:38:59 [22304] ERROR:core:forward_reply: no 2nd via found in 
> reply
> Dec 26 21:39:00 [22303] ERROR:core:forward_reply: no 2nd via found in 
> reply
> Dec 26 21:39:10 [22308] ERROR:core:forward_reply: no 2nd via found in 
> reply
> Dec 26 21:39:11 [22306] ERROR:core:forward_reply: no 2nd via found in 
> reply
> D
>
> -- 
>
>
> My opensips.cfg is as follows:
>
> route{
>
>     # initial sanity checks -- messages with
>     # max_forwards==0, or excessively long requests
>
>     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;
>     };
>
>     # we record-route all messages -- to make sure that
>     # subsequent messages will go through our proxy; that's
>     # particularly good if upstream and downstream entities
>     # use different transport protocol
>
>     if (!method=="REGISTER")
>         record_route();
>
>     # subsequent messages withing a dialog should take the
>     # path determined by record-routing
>
>     if (loose_route()) {
>         # mark routing logic in request
>         append_hf("P-hint: rr-enforced\r\n");
>         route(1);
>     };
>
>     if (!uri==myself) {
>         # mark routing logic in request
>         append_hf("P-hint: outbound\r\n");
>         route(1);
>     };
>
>     # if the request is for other domain use UsrLoc
>     # (in case, it does not work, use the following command
>     # with proper names and addresses in it)
>     if (uri==myself) {
>
>         if (method=="REGISTER") {
>             if (!www_authorize("", "subscriber")) {
>                 www_challenge("", "0");
>                 exit;
>             };
>
>             save("location");
>             exit;
>         };
>
>         # requests for Media server
>         if(is_method("INVITE") && !has_totag() && uri=~"sip:\*9") {
>             route(3);
>             exit;
>         }
>
>         # mark transaction if user is in voicemail group
>         if(is_method("INVITE") && !has_totag()
>             && is_user_in("Request-URI","voicemail"))
>         {
>             xdbg("user [$ru] has voicemail redirection enabled\n");
>             # backup R-URI
>             avp_pushto("$ru","$avp(i:10)");
>             #avp_write("$ruri","$avp(i:10)");
>             setflag(2);
>         };
>         # native SIP destinations are handled using our USRLOC DB
>         if (!lookup("location")) {
>             if(isflagset(2)) {
>                 # route to Asterisk Media Server
>                 prefix("1");
>                 rewritehostport("192.168.1.11:5060 
> <http://192.168.1.11:5060>");
>                 route(1);
>             } else {
>                 sl_send_reply("404", "Not Found");
>                 exit;
>             }
>         };
>         append_hf("P-hint: usrloc applied\r\n");
>     };
>
>     route(1);
> }
>
>
> route[1] {
>    
>     if(isflagset(2))
>         t_on_failure("1");
>
>     if (!t_relay()) {
>         sl_reply_error();
>     };
>     exit;
> }
>
>
> # voicemail access
> # - *98 - listen caller's voice messages, being prompted for pin
> # - *981 - listen voice messages, being promted for mailbox and pin
> # - *98XXXX - leave voice message to XXXX
> #
> route[3] {
>       # direct voicemail
>     if (uri =~ "sip:\*98@" ) {
>             rewriteuser("1");
>         xdbg("voicemail access\n");
>     } else if (uri =~ "sip:\*981@" ) {
>          strip(4);
>         rewriteuser("11");
>     } else if (uri =~ "sip:\*98.+@" ) {
>          strip(3);
>         prefix("1");
>     } else {
>         xlog("unknown media extension $rU\n");
>         sl_send_reply("404", "Unknown media service");
>         exit;
>     }
>
>     # route to Asterisk Media Server
>     rewritehostport("192.168.1.11:5060 <http://192.168.1.11:5060>");
>     route(1);
> }
>
> failure_route[1] {
>     if (t_was_cancelled()) {
>         xdbg("transaction was cancelled by UAC\n");
>         return;
>     }
>     # restore initial uri
>     avp_pushto("$ru","$avp(i:10)");
>     #avp_pushto("$ru", "i:10");
>     prefix("1");
>     # route to Asterisk Media Server
>     rewritehostport("192.168.1.11:5060 <http://192.168.1.11:5060>");
>     resetflag(2);
>     route(1);
> }
>
>
> Thank you,
>
>
> Khan
>
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>
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