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.

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

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(a)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/ >