Hello Daniel

I do have quite a few core files, please send me the gdb commands.

Regards

Panagiotis.

Daniel-Constantin Mierla wrote:

Hello Panagiotis,

On 03/16/2010 02:21 PM, Panagiotis Skoulikaritis wrote:

...

Hello Daniel

I will need time to recreate the problem, attached are the only traces I kept. on the amaze-4.cap the calls are originated by a softphone registered on the kamailio on the crash.cap the calls are originated from the "PSTN".

we do account the $pd and the $pn

I reply to you directly since i don't want to give the traces on the mailing list, I hope you will understand.

I asked to be sent private, it is ok.

...

Also for workaround Alex took out the PDT module from the route and the kamailio do not crash anymore.

Hmm, so you say pdt module is related? That is pretty small and old module...

Do you still have the core file? I can send you some gdb commands to get more details out of it.

Thanks, Daniel

...

Regards

Panagiotis

Daniel-Constantin Mierla wrote:

...

Hello,

can you send me ngrep/pcap file with ip addresses so I can match which 200ok is causing the problem (coming from B or coming from Asterisk)? The backtrace shows ip while the sip trace is masked.

Also, I would need a bit more info from the core file. Please keep one around. The issue seems to be related to P-Asserted-Identity header, but I couldn't find any such header in the sip trace you sent.

Are you accounting the PAI header?

Thanks, Daniel

On 03/10/2010 03:30 PM, Panagiotis Skoulikaritis wrote:

...

Dear Marius

The scenario is as follows:

1. A Call is placed by a sip subscriber "A"
2. kamailio forwards the call to the asterisk server
3. Asterisk plays an IVR message on the subscriber "A", creates a

new call to a "virtual" number which is forwarded to the kamailio server, and plays an ivr to this leg as well when the call is answered, then it connects the two calls. 4. Kamailio translates the "virtual" number to the pstn number of subscriber B

I have attached a picture of the above scenario.

The modules that are loaded are:

loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "mi_datagram.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "pdt.so" loadmodule "dialplan.so" loadmodule "siptrace.so" loadmodule "dialog.so" loadmodule "sqlops.so" loadmodule "userblacklist.so" loadmodule "htable.so" loadmodule "uac.so"

The config that does all the routing is :

route[10] {

   xlog("alx ------- This is Route 10 -------");

   if($rU =~ "^.*%+")
   {
           xlog("alx ------- The number contains %23 ");
           $rU = $(rU{re.subst,/^(.*)%23(.*)/\1\2/});
           #$rU = $(rU{s.unescape.user}); #It changes the %23 

to # !! xlog("alx ------- The perl $rU ------- "); }

   if($rU =~ "^.*#+")
   {
           xlog("alx ------- The number contains #");
           $rU = $(rU{re.subst,/^(.*)#(.*)/\1\2/});
           #$rU = $(rU{s.unescape.user}); #It changes the %23 

to # !! xlog("alx ------- The perl $rU ------- "); }

   if(prefix2domain("2", "0")) {

           $var(dial_grp) = $(rd{s.select,0,.}{s.int}); # 

Dialplan group prefix for routing $var(num_pr) = $(rd{s.select,1,.}{s.int}); # The number of digits that prefix has $var(num_translation) = $(rd{s.select,2,.}{s.int}); # Called number translation $avp(s:port_translation) = $(rd{s.select,3,.}{s.int}); # Port number translation #$var(test_var) = $(rd{s.select,4,.}{s.int}); # Future property

           $avp(s:cust_prefix) = $(rU{s.substr,0,$var(num_pr)});
           $rU = $(rU{s.substr,$var(num_pr),0});

   xlog("alx ------- The new rU is $rU and properties $rd 

-------");

   if($var(num_translation) == 1)
   {
           if($sht(a=>$rU)!=null){

                 $rU = $sht(a=>$rU);
                 xlog("alx ------- Translation Done. DST 

num=$rU ----------");

           } else {
                   xlog("alx ------- Translation NOT Done  

----------"); }

                   #xlog("alx ------- We have DST number 

translation for user fU $avp(s:frm_user_name) ----------"); #if(dp_translate("31", "$rU/$rU")) #{ # xlog("alx ------- Translation Done. DST num=$rU ----------"); #} else { # xlog("alx ------- Translation NOT Done ----------"); #} }

   if(dp_translate("$var(dial_grp)", "$rU/$rU"))
           {
                   xlog("alx ------- The $rU and with 

attributes :$avp(s:dest) -------\n");

                   $var(i) = 0;
                   while($(avp(s:dest){s.select,$var(i),.})!="#")
                   {
                           $avp(s:dstgrp) = 

$(avp(s:dest){s.select,$var(i),.}{s.int}); $var(i) = $var(i) + 1; xlog("alx ------- The avp(s:dstgrp)=$avp(s:dstgrp) var(i)=$var(i) -------"); }

                   # backup the username so we can use 

different prefixes $avp(s:user) = $rU;

                   # select destination from first group

if(ds_select_domain("$avp(s:dstgrp)", "4")) {

if($(ru{uri.param,prefix})!=null) { $ru = "sip:" + $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd;

                                           } else {
                                                           $ru 

= "sip:" + $avp(s:user) + "@" + $rd; } }

                           $avp(s:dstgrp) = null;
                                   xlog("alx ------- The final 

RURI is $ru ------- "); if($avp(s:port_translation) == 1) { rewriteport("5061"); } t_on_failure("3"); t_relay(); exit;

           }




   }

}

Attached is the trace

Regards.

P.

marius zbihlei wrote:

...

Panagiotis Skoulikaritis wrote:

...

Hello Daniel

the kamailio version is 1.5.3

Regards

P.

Hello,

Can you give us more details like the sip message that generates the coredump (or if every sip message received generates the core), if your config does something more out of the ordinary(let's say exotic). Can we reproduce it ?

It would also be helpful if you specify the list of modules you have loaded.

Cheers, Marius

...

Daniel-Constantin Mierla wrote: > Hello, http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

---

Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

-- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010

• http://www.asipto.com/index.php/sip-router-masterclass/

-- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010

• http://www.asipto.com/index.php/sip-router-masterclass/