[OpenSIPS-Users] OpenSIPS is not running, Erorr
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Fri Jan 9 12:51:08 CET 2009
Khan,
Have you identify the SIP reply that generated this error ?
Regards,
Bogdan
Khan Friend wrote:
> Bogdan,
>
> The problem is that I don't know much about SIP server and VoIP. This
> is experimental project, I studied and successfully ran simple
> OpenSIPS server. When I try to add Asterisk or NAT Traversal, I ran
> into many problems. One of them is this (Asterisk config), I traced
> the log file but not much luck understanding what part needs fixing.
>
> Please help me identify the root of the problem and how to fix. How do
> i find SIP replies, what do i do to see them and capture them.
>
>
> Thanks in advance,
>
> Khan
>
> On Sun, Dec 28, 2008 at 4:33 AM, Bogdan-Andrei Iancu
> <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:
>
> 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> <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 $rU\n");
> sl_send_reply("404", "Unknown media service");
> exit;
> }
>
> # route to Asterisk Media Server
> rewritehostport("192.168.1.11:5060
> <http://192.168.1.11:5060> <http://192.168.1.11:5060>");
>
> route(1);
> }
>
> failure_route[1] {
> if (t_was_cancelled()) {
> xdbg("transaction was cancelled by UAC\n");
> return;
> }
> # restore initial uri
> avp_pushto("$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> <http://192.168.1.11:5060>");
>
> resetflag(2);
> route(1);
> }
>
>
> Thank you,
>
>
> Khan
>
> ------------------------------------------------------------------------
>
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>
>
> --
> Thank you,
>
>
> Mr. Khan
> Director Technical Resources, Research, and Deployment.
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