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Hello Daniel<br>
<br>
I do have quite a few core files, please send me the gdb commands.<br>
<br>
Regards<br>
<br>
Panagiotis.<br>
<br>
Daniel-Constantin Mierla wrote:
<blockquote cite="mid:4B9F8D0E.3090601@gmail.com" type="cite">
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Hello Panagiotis,<br>
<br>
On 03/16/2010 02:21 PM, Panagiotis Skoulikaritis wrote:
<blockquote cite="mid:4B9F85EB.6060403@algonet.gr" type="cite">
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Hello Daniel<br>
<br>
I will need time to recreate the problem,<br>
attached are the only traces I kept.<br>
on the amaze-4.cap the calls are originated by a softphone registered
on the kamailio<br>
on the crash.cap the calls are originated from the "PSTN".<br>
<br>
we do account the $pd and the $pn<br>
<br>
I reply to you directly since i don't want to give the traces on the
mailing list, I hope you will understand.<br>
</blockquote>
<br>
I asked to be sent private, it is ok.<br>
<br>
<blockquote cite="mid:4B9F85EB.6060403@algonet.gr" type="cite"><br>
Also for workaround Alex took out the PDT module from the route and the
kamailio do not crash anymore.<br>
</blockquote>
<br>
Hmm, so you say pdt module is related? That is pretty small and old
module...<br>
<br>
Do you still have the core file? I can send you some gdb commands to
get more details out of it.<br>
<br>
Thanks,<br>
Daniel<br>
<br>
<blockquote cite="mid:4B9F85EB.6060403@algonet.gr" type="cite"><br>
Regards<br>
<br>
Panagiotis<br>
<br>
<br>
Daniel-Constantin Mierla wrote:
<blockquote cite="mid:4B9D03D6.8060000@gmail.com" type="cite">
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<title></title>
Hello,<br>
<br>
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.<br>
<br>
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.<br>
<br>
Are you accounting the PAI header?<br>
<br>
Thanks,<br>
Daniel<br>
<br>
On 03/10/2010 03:30 PM, Panagiotis Skoulikaritis wrote:
<blockquote cite="mid:4B97ACF3.80609@algonet.gr" type="cite">Dear
Marius <br>
<br>
The scenario is as follows: <br>
<br>
1. A Call is placed by a sip subscriber "A" <br>
2. kamailio forwards the call to the asterisk server <br>
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. <br>
4. Kamailio translates the "virtual" number to the pstn number of
subscriber B <br>
<br>
<br>
I have attached a picture of the above scenario. <br>
<br>
The modules that are loaded are: <br>
<br>
loadmodule "db_mysql.so" <br>
loadmodule "mi_fifo.so" <br>
loadmodule "mi_datagram.so" <br>
loadmodule "sl.so" <br>
loadmodule "tm.so" <br>
loadmodule "rr.so" <br>
loadmodule "pv.so" <br>
loadmodule "maxfwd.so" <br>
loadmodule "usrloc.so" <br>
loadmodule "registrar.so" <br>
loadmodule "textops.so" <br>
loadmodule "uri_db.so" <br>
loadmodule "siputils.so" <br>
loadmodule "xlog.so" <br>
loadmodule "acc.so" <br>
loadmodule "dispatcher.so" <br>
loadmodule "pdt.so" <br>
loadmodule "dialplan.so" <br>
loadmodule "siptrace.so" <br>
loadmodule "dialog.so" <br>
loadmodule "sqlops.so" <br>
loadmodule "userblacklist.so" <br>
loadmodule "htable.so" <br>
loadmodule "uac.so" <br>
<br>
<br>
The config that does all the routing is : <br>
<br>
route[10] { <br>
<br>
xlog("alx ------- This is Route 10 -------"); <br>
<br>
if($rU =~ "^.*%+") <br>
{ <br>
xlog("alx ------- The number contains %23 "); <br>
$rU = $(rU{re.subst,/^(.*)%23(.*)/\1\2/}); <br>
#$rU = $(rU{s.unescape.user}); #It changes the %23 to #
!! <br>
xlog("alx ------- The perl $rU ------- "); <br>
} <br>
<br>
if($rU =~ "^.*#+") <br>
{ <br>
xlog("alx ------- The number contains #"); <br>
$rU = $(rU{re.subst,/^(.*)#(.*)/\1\2/}); <br>
#$rU = $(rU{s.unescape.user}); #It changes the %23 to #
!! <br>
xlog("alx ------- The perl $rU ------- "); <br>
} <br>
<br>
if(prefix2domain("2", "0")) { <br>
<br>
$var(dial_grp) = $(rd{s.select,0,.}{s.int}); # Dialplan
group prefix for routing <br>
$var(num_pr) = $(rd{s.select,1,.}{s.int}); # The number
of digits that prefix has <br>
$var(num_translation) = $(rd{s.select,2,.}{s.int}); #
Called number translation <br>
$avp(s:port_translation) = $(rd{s.select,3,.}{s.int});
# Port number translation <br>
#$var(test_var) = $(rd{s.select,4,.}{s.int}); # Future
property <br>
<br>
$avp(s:cust_prefix) = $(rU{s.substr,0,$var(num_pr)}); <br>
$rU = $(rU{s.substr,$var(num_pr),0}); <br>
<br>
xlog("alx ------- The new rU is $rU and properties $rd
-------"); <br>
<br>
if($var(num_translation) == 1) <br>
{ <br>
if($sht(a=>$rU)!=null){ <br>
<br>
$rU = $sht(a=>$rU); <br>
xlog("alx ------- Translation Done. DST num=$rU
----------"); <br>
<br>
} else { <br>
xlog("alx ------- Translation NOT Done
----------"); <br>
} <br>
<br>
<br>
#xlog("alx ------- We have DST number
translation for user fU $avp(s:frm_user_name) ----------"); <br>
#if(dp_translate("31", "$rU/$rU")) <br>
#{ <br>
# xlog("alx ------- Translation Done. DST
num=$rU ----------"); <br>
#} else { <br>
# xlog("alx ------- Translation NOT Done
----------"); <br>
#} <br>
} <br>
<br>
if(dp_translate("$var(dial_grp)", "$rU/$rU")) <br>
{ <br>
xlog("alx ------- The $rU and with attributes
:$avp(s:dest) -------\n"); <br>
<br>
$var(i) = 0; <br>
while($(avp(s:dest){s.select,$var(i),.})!="#") <br>
{ <br>
$avp(s:dstgrp) =
$(avp(s:dest){s.select,$var(i),.}{s.int}); <br>
$var(i) = $var(i) + 1; <br>
xlog("alx ------- The
avp(s:dstgrp)=$avp(s:dstgrp) var(i)=$var(i) -------"); <br>
} <br>
<br>
# backup the username so we can use different
prefixes <br>
$avp(s:user) = $rU; <br>
<br>
# select destination from first group <br>
if(ds_select_domain("$avp(s:dstgrp)",
"4")) <br>
{ <br>
if($(ru{uri.param,prefix})!=null) <br>
{ <br>
$ru =
"sip:" + $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; <br>
<br>
} else { <br>
$ru =
"sip:" + $avp(s:user) + "@" + $rd; <br>
} <br>
} <br>
<br>
$avp(s:dstgrp) = null; <br>
xlog("alx ------- The final RURI
is $ru ------- "); <br>
if($avp(s:port_translation) ==
1) <br>
{ <br>
rewriteport("5061"); <br>
} <br>
t_on_failure("3"); <br>
t_relay(); <br>
exit; <br>
<br>
} <br>
<br>
<br>
<br>
<br>
} <br>
<br>
<br>
} <br>
<br>
Attached is the trace <br>
<br>
Regards. <br>
<br>
P. <br>
<br>
marius zbihlei wrote: <br>
<blockquote type="cite">Panagiotis Skoulikaritis wrote: <br>
<blockquote type="cite">Hello Daniel <br>
<br>
the kamailio version is 1.5.3 <br>
<br>
Regards <br>
<br>
P. <br>
</blockquote>
Hello, <br>
<br>
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 ? <br>
<br>
It would also be helpful if you specify the list of modules you have
loaded. <br>
<br>
Cheers, <br>
Marius <br>
<blockquote type="cite"><br>
Daniel-Constantin Mierla wrote: <br>
<blockquote type="cite">Hello, <br>
</blockquote>
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</blockquote>
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<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
* <a moz-do-not-send="true" class="moz-txt-link-freetext"
href="http://www.asipto.com/index.php/sip-router-masterclass/">http://www.asipto.com/index.php/sip-router-masterclass/</a>
</pre>
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</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
* <a moz-do-not-send="true" class="moz-txt-link-freetext"
href="http://www.asipto.com/index.php/sip-router-masterclass/">http://www.asipto.com/index.php/sip-router-masterclass/</a>
</pre>
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