Dear fellow kaimailio users
We have a kamailio server which crashes. below is the backtrace from the core files any idea why the kamailio is crashing
Regards
Panagiotis
core.29568 Mar 10 09:27
#0 fm_status (qm=0x73a040) at mem/f_malloc.c:609 #1 0x0000000000423d5c in sig_usr (signo=15) at main.c:563 #2 <signal handler called> #3 0x00000037e3cd4711 in __recvfrom_nocancel () from /lib64/libc.so.6 #4 0x00000000004790cc in udp_rcv_loop () at udp_server.c:408 #5 0x000000000042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774
core.19350 Mar 10 09:05
(gdb) backtrace #0 free_to (tb=0x775c00) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2ad2432de100) at parser/hf.c:187 #2 0x00002ad23fe3e525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2ad2432dcf58, rpl=0x772d28, code=<value optimized out>) at sip_msg.h:54 #3 0x00002ad23fe47b46 in t_reply_matching (p_msg=0x772d28, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002ad23fe47fa2 in t_check (p_msg=0x772d28, param_branch=0x7ffff9c016bc) at t_lookup.c:964 #5 0x00002ad23fe58ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x772d28) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bK45f7.70f91294.0;received=77.247.97.11\r\nV", len=920, rcv_info=0x7ffff9c017a0) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=3, argv=0x7ffff9c019b8) at main.c:774 (gdb)
core.29567 Mar 10 09:27
gdb) backtrace #0 free_to (tb=0x776460) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2ad1805fa100) at parser/hf.c:187 #2 0x00002ad17d15a525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2ad1805f8f58, rpl=0x771920, code=<value optimized out>) at sip_msg.h:54 #3 0x00002ad17d163b46 in t_reply_matching (p_msg=0x771920, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002ad17d163fa2 in t_check (p_msg=0x771920, param_branch=0x7fff8bf6a8cc) at t_lookup.c:964 #5 0x00002ad17d174ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x771920) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bKddc9.fa58f7e.0;received=77.247.97.11\r\nVia: SIP/2.0/UDP 213.170.194.47:5060;branch=z9hG4bKb973f6a69c9bea270e9db867dd7cc90f\r\nRecord-Route: <si"..., len=919, rcv_info=0x7fff8bf6a9b0) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774 (gdb)
core.5326 Mar 10 09:02
(gdb) backtrace #0 free_to (tb=0x772c48) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2b2578e22560) at parser/hf.c:187 #2 0x00002b257597b525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2b2578e213b8, rpl=0x771b50, code=<value optimized out>) at sip_msg.h:54 #3 0x00002b2575984b46 in t_reply_matching (p_msg=0x771b50, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002b2575984fa2 in t_check (p_msg=0x771b50, param_branch=0x7fff7cdc63ac) at t_lookup.c:964 #5 0x00002b2575995ac2 in reply_received (p_msg=0x772c48) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x771b50) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bKec2f.dc3d97a7.0;received=77.247.97.11\r\nVia: SIP/2.0/UDP 213.170.194.47:5060;branch=z9hG4bK47703457194f5be415efc231f6b3e923\r\nRecord-Route: <s"..., len=918, rcv_info=0x7fff7cdc6490) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=5, argv=0x7fff7cdc66a8) at main.c:774 (gdb) quit
Hello,
what version do you have? If it is for 3.0, please register a bug at: http://sip-router.org/tracker/
In 3.0 the crash is at:
186 case HDR_REFER_TO_T: 187 free_to(hf->parsed); 188 break;
I am out of the office without my linux box these days to be able to check more. Maybe some other devel can look a bit at it.
Thanks, Daniel
On Wed, Mar 10, 2010 at 9:17 AM, Panagiotis Skoulikaritis <pskoul@algonet.gr
wrote:
Dear fellow kaimailio users
We have a kamailio server which crashes. below is the backtrace from the core files any idea why the kamailio is crashing
Regards
Panagiotis
core.29568 Mar 10 09:27
#0 fm_status (qm=0x73a040) at mem/f_malloc.c:609 #1 0x0000000000423d5c in sig_usr (signo=15) at main.c:563 #2 <signal handler called> #3 0x00000037e3cd4711 in __recvfrom_nocancel () from /lib64/libc.so.6 #4 0x00000000004790cc in udp_rcv_loop () at udp_server.c:408 #5 0x000000000042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774
core.19350 Mar 10 09:05
(gdb) backtrace #0 free_to (tb=0x775c00) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2ad2432de100) at parser/hf.c:187 #2 0x00002ad23fe3e525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2ad2432dcf58, rpl=0x772d28, code=<value optimized out>) at sip_msg.h:54 #3 0x00002ad23fe47b46 in t_reply_matching (p_msg=0x772d28, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002ad23fe47fa2 in t_check (p_msg=0x772d28, param_branch=0x7ffff9c016bc) at t_lookup.c:964 #5 0x00002ad23fe58ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x772d28) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bK45f7.70f91294.0;received=77.247.97.11\r\nV", len=920, rcv_info=0x7ffff9c017a0) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=3, argv=0x7ffff9c019b8) at main.c:774 (gdb)
core.29567 Mar 10 09:27
gdb) backtrace #0 free_to (tb=0x776460) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2ad1805fa100) at parser/hf.c:187 #2 0x00002ad17d15a525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2ad1805f8f58, rpl=0x771920, code=<value optimized out>) at sip_msg.h:54 #3 0x00002ad17d163b46 in t_reply_matching (p_msg=0x771920, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002ad17d163fa2 in t_check (p_msg=0x771920, param_branch=0x7fff8bf6a8cc) at t_lookup.c:964 #5 0x00002ad17d174ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x771920) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bKddc9.fa58f7e.0;received=77.247.97.11\r\nVia: SIP/2.0/UDP 213.170.194.47:5060;branch=z9hG4bKb973f6a69c9bea270e9db867dd7cc90f\r\nRecord-Route: <si"..., len=919, rcv_info=0x7fff8bf6a9b0) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774 (gdb)
core.5326 Mar 10 09:02
(gdb) backtrace #0 free_to (tb=0x772c48) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2b2578e22560) at parser/hf.c:187 #2 0x00002b257597b525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2b2578e213b8, rpl=0x771b50, code=<value optimized out>) at sip_msg.h:54 #3 0x00002b2575984b46 in t_reply_matching (p_msg=0x771b50, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002b2575984fa2 in t_check (p_msg=0x771b50, param_branch=0x7fff7cdc63ac) at t_lookup.c:964 #5 0x00002b2575995ac2 in reply_received (p_msg=0x772c48) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x771b50) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bKec2f.dc3d97a7.0;received=77.247.97.11\r\nVia: SIP/2.0/UDP 213.170.194.47:5060;branch=z9hG4bK47703457194f5be415efc231f6b3e923\r\nRecord-Route: <s"..., len=918, rcv_info=0x7fff7cdc6490) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=5, argv=0x7fff7cdc66a8) at main.c:774 (gdb) quit
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
Hello Daniel
the kamailio version is 1.5.3
Regards
P.
Daniel-Constantin Mierla wrote:
Hello,
what version do you have? If it is for 3.0, please register a bug at: http://sip-router.org/tracker/
In 3.0 the crash is at:
186 case HDR_REFER_TO_T: 187 free_to(hf->parsed); 188 break;
I am out of the office without my linux box these days to be able to check more. Maybe some other devel can look a bit at it.
Thanks, Daniel
On Wed, Mar 10, 2010 at 9:17 AM, Panagiotis Skoulikaritis <pskoul@algonet.gr mailto:pskoul@algonet.gr> wrote:
Dear fellow kaimailio users We have a kamailio server which crashes. below is the backtrace from the core files any idea why the kamailio is crashing Regards Panagiotis core.29568 Mar 10 09:27 #0 fm_status (qm=0x73a040) at mem/f_malloc.c:609 #1 0x0000000000423d5c in sig_usr (signo=15) at main.c:563 #2 <signal handler called> #3 0x00000037e3cd4711 in __recvfrom_nocancel () from /lib64/libc.so.6 #4 0x00000000004790cc in udp_rcv_loop () at udp_server.c:408 #5 0x000000000042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774 core.19350 Mar 10 09:05 (gdb) backtrace #0 free_to (tb=0x775c00) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2ad2432de100) at parser/hf.c:187 #2 0x00002ad23fe3e525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2ad2432dcf58, rpl=0x772d28, code=<value optimized out>) at sip_msg.h:54 #3 0x00002ad23fe47b46 in t_reply_matching (p_msg=0x772d28, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002ad23fe47fa2 in t_check (p_msg=0x772d28, param_branch=0x7ffff9c016bc) at t_lookup.c:964 #5 0x00002ad23fe58ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x772d28) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bK45f7.70f91294.0;received=77.247.97.11\r\nV", len=920, rcv_info=0x7ffff9c017a0) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=3, argv=0x7ffff9c019b8) at main.c:774 (gdb) core.29567 Mar 10 09:27 gdb) backtrace #0 free_to (tb=0x776460) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2ad1805fa100) at parser/hf.c:187 #2 0x00002ad17d15a525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2ad1805f8f58, rpl=0x771920, code=<value optimized out>) at sip_msg.h:54 #3 0x00002ad17d163b46 in t_reply_matching (p_msg=0x771920, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002ad17d163fa2 in t_check (p_msg=0x771920, param_branch=0x7fff8bf6a8cc) at t_lookup.c:964 #5 0x00002ad17d174ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x771920) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bKddc9.fa58f7e.0;received=77.247.97.11\r\nVia: SIP/2.0/UDP 213.170.194.47:5060;branch=z9hG4bKb973f6a69c9bea270e9db867dd7cc90f\r\nRecord-Route: <si"..., len=919, rcv_info=0x7fff8bf6a9b0) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774 (gdb) core.5326 Mar 10 09:02 (gdb) backtrace #0 free_to (tb=0x772c48) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2b2578e22560) at parser/hf.c:187 #2 0x00002b257597b525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2b2578e213b8, rpl=0x771b50, code=<value optimized out>) at sip_msg.h:54 #3 0x00002b2575984b46 in t_reply_matching (p_msg=0x771b50, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002b2575984fa2 in t_check (p_msg=0x771b50, param_branch=0x7fff7cdc63ac) at t_lookup.c:964 #5 0x00002b2575995ac2 in reply_received (p_msg=0x772c48) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x771b50) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bKec2f.dc3d97a7.0;received=77.247.97.11\r\nVia: SIP/2.0/UDP 213.170.194.47:5060;branch=z9hG4bK47703457194f5be415efc231f6b3e923\r\nRecord-Route: <s"..., len=918, rcv_info=0x7fff7cdc6490) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=5, argv=0x7fff7cdc66a8) at main.c:774 (gdb) quit _______________________________________________ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org <mailto: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 http://www.asipto.com
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
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
No. Time Source Destination Protocol Info 11 3.576857 sip subscriber A kamailio SIP/SDP Request: INVITE sip:virtual number@kamailio;transport=UDP, with session description
Internet Protocol, Src: sip subscriber A (sip subscriber A), Dst: kamailio (kamailio) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:virtual number@kamailio;transport=UDP SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Via: SIP/2.0/UDP sip subscriber A:5060;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport Transport: UDP Sent-by Address: sip subscriber A Sent-by port: 5060 Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z- RPort: rport Max-Forwards: 70 Contact: <sip:sip subscriber A@sip subscriber A:5060> Contact Binding: <sip:sip subscriber A@sip subscriber A:5060> URI: <sip:sip subscriber A@sip subscriber A:5060> SIP contact address: sip:sip subscriber A@sip subscriber A:5060 To: <sip:virtual number@kamailio>;transport=UDP SIP to address: sip:virtual number@kamailio From: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@kamailio SIP tag: fe71df57 Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 1 INVITE Sequence Number: 1 Method: INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO Content-Type: application/sdp User-Agent: Zoiper rev.448 Content-Length: 245 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): Zoiper_user 0 0 IN IP4 10.0.2.46 Owner Username: Zoiper_user Session ID: 0 Session Version: 0 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 10.0.2.46 Session Name (s): Zoiper_user Connection Information (c): IN IP4 sip subscriber A Connection Network Type: IN Connection Address Type: IP4 Connection Address: sip subscriber A Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 Media Description, name and address (m): audio 8000 RTP/AVP 8 0 3 101 Media Type: audio Media Port: 8000 Media Proto: RTP/AVP Media Format: ITU-T G.711 PCMA Media Format: ITU-T G.711 PCMU Media Format: GSM 06.10 Media Format: 101 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute Fieldname: rtpmap Media Format: 8 MIME Type: PCMA Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Format: 0 MIME Type: PCMU Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute Fieldname: rtpmap Media Format: 3 MIME Type: GSM Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Format: 101 MIME Type: telephone-event Media Attribute (a): fmtp:101 0-15 Media Attribute Fieldname: fmtp Media Format: 101 [telephone-event] Media format specific parameters: 0-15 Media Attribute (a): sendrecv
No. Time Source Destination Protocol Info 12 3.577798 kamailio sip subscriber A SIP Status: 100 Giving a try
Internet Protocol, Src: kamailio (kamailio), Dst: sip subscriber A (sip subscriber A) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Giving a try Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP sip subscriber A:5060;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060 Transport: UDP Sent-by Address: sip subscriber A Sent-by port: 5060 Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z- RPort: 5060 To: <sip:virtual number@kamailio>;transport=UDP SIP to address: sip:virtual number@kamailio From: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@kamailio SIP tag: fe71df57 Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 1 INVITE Sequence Number: 1 Method: INVITE Server: Kamailio (1.5.3-notls (x86_64/linux)) Content-Length: 0
No. Time Source Destination Protocol Info 13 3.577893 kamailio asterisk SIP/SDP Request: INVITE sip:virtual number@asterisk, with session description
Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:virtual number@asterisk SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Record-Route: sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887 Via: SIP/2.0/UDP kamailio;branch=z9hG4bKce3c.92b7a847.0 Transport: UDP Sent-by Address: kamailio Branch: z9hG4bKce3c.92b7a847.0 Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber A;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060 Transport: UDP Sent-by Address: sip subscriber A Sent-by port: 5060 Received: sip subscriber A Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z- RPort: 5060 Max-Forwards: 69 Contact: <sip:sip subscriber A@sip subscriber A:5060> Contact Binding: <sip:sip subscriber A@sip subscriber A:5060> URI: <sip:sip subscriber A@sip subscriber A:5060> SIP contact address: sip:sip subscriber A@sip subscriber A:5060 To: <sip:virtual number@kamailio>;transport=UDP SIP to address: sip:virtual number@kamailio From: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@kamailio SIP tag: fe71df57 Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 1 INVITE Sequence Number: 1 Method: INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO Content-Type: application/sdp User-Agent: Zoiper rev.448 Content-Length: 245 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): Zoiper_user 0 0 IN IP4 10.0.2.46 Owner Username: Zoiper_user Session ID: 0 Session Version: 0 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 10.0.2.46 Session Name (s): Zoiper_user Connection Information (c): IN IP4 sip subscriber A Connection Network Type: IN Connection Address Type: IP4 Connection Address: sip subscriber A Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 Media Description, name and address (m): audio 8000 RTP/AVP 8 0 3 101 Media Type: audio Media Port: 8000 Media Proto: RTP/AVP Media Format: ITU-T G.711 PCMA Media Format: ITU-T G.711 PCMU Media Format: GSM 06.10 Media Format: 101 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute Fieldname: rtpmap Media Format: 8 MIME Type: PCMA Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Format: 0 MIME Type: PCMU Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute Fieldname: rtpmap Media Format: 3 MIME Type: GSM Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Format: 101 MIME Type: telephone-event Media Attribute (a): fmtp:101 0-15 Media Attribute Fieldname: fmtp Media Format: 101 [telephone-event] Media format specific parameters: 0-15 Media Attribute (a): sendrecv
No. Time Source Destination Protocol Info 14 3.578452 asterisk kamailio SIP Status: 100 Trying
Internet Protocol, Src: asterisk (asterisk), Dst: kamailio (kamailio) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP kamailio;branch=z9hG4bKce3c.92b7a847.0;received=kamailio Transport: UDP Sent-by Address: kamailio Branch: z9hG4bKce3c.92b7a847.0 Received: kamailio Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber A;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060 Transport: UDP Sent-by Address: sip subscriber A Sent-by port: 5060 Received: sip subscriber A Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z- RPort: 5060 Record-Route: sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887 From: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@kamailio SIP tag: fe71df57 To: <sip:virtual number@kamailio>;transport=UDP SIP to address: sip:virtual number@kamailio Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 1 INVITE Sequence Number: 1 Method: INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:virtual number@asterisk> Contact Binding: <sip:virtual number@asterisk> URI: <sip:virtual number@asterisk> SIP contact address: sip:virtual number@asterisk Content-Length: 0
No. Time Source Destination Protocol Info 15 3.578736 asterisk kamailio SIP/SDP Status: 183 Session Progress, with session description
Internet Protocol, Src: asterisk (asterisk), Dst: kamailio (kamailio) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 183 Session Progress Status-Code: 183 [Resent Packet: False] Message Header Via: SIP/2.0/UDP kamailio;branch=z9hG4bKce3c.92b7a847.0;received=kamailio Transport: UDP Sent-by Address: kamailio Branch: z9hG4bKce3c.92b7a847.0 Received: kamailio Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber A;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060 Transport: UDP Sent-by Address: sip subscriber A Sent-by port: 5060 Received: sip subscriber A Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z- RPort: 5060 Record-Route: sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887 From: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@kamailio SIP tag: fe71df57 To: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c SIP to address: sip:virtual number@kamailio SIP tag: as416d507c Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 1 INVITE Sequence Number: 1 Method: INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:virtual number@asterisk> Contact Binding: <sip:virtual number@asterisk> URI: <sip:virtual number@asterisk> SIP contact address: sip:virtual number@asterisk Content-Type: application/sdp Content-Length: 285 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 4184 4184 IN IP4 asterisk Owner Username: root Session ID: 4184 Session Version: 4184 Owner Network Type: IN Owner Address Type: IP4 Owner Address: asterisk Session Name (s): session Connection Information (c): IN IP4 asterisk Connection Network Type: IN Connection Address Type: IP4 Connection Address: asterisk Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 Media Description, name and address (m): audio 16620 RTP/AVP 3 0 8 101 Media Type: audio Media Port: 16620 Media Proto: RTP/AVP Media Format: GSM 06.10 Media Format: ITU-T G.711 PCMU Media Format: ITU-T G.711 PCMA Media Format: 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute Fieldname: rtpmap Media Format: 3 MIME Type: GSM Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Format: 0 MIME Type: PCMU Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute Fieldname: rtpmap Media Format: 8 MIME Type: PCMA Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Format: 101 MIME Type: telephone-event Media Attribute (a): fmtp:101 0-16 Media Attribute Fieldname: fmtp Media Format: 101 [telephone-event] Media format specific parameters: 0-16 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - Media Attribute (a): ptime:20 Media Attribute Fieldname: ptime Media Attribute Value: 20 Media Attribute (a): sendrecv
No. Time Source Destination Protocol Info 16 3.578818 asterisk kamailio SIP/SDP Status: 200 OK, with session description
Internet Protocol, Src: asterisk (asterisk), Dst: kamailio (kamailio) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP kamailio;branch=z9hG4bKce3c.92b7a847.0;received=kamailio Transport: UDP Sent-by Address: kamailio Branch: z9hG4bKce3c.92b7a847.0 Received: kamailio Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber A;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060 Transport: UDP Sent-by Address: sip subscriber A Sent-by port: 5060 Received: sip subscriber A Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z- RPort: 5060 Record-Route: sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887 From: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@kamailio SIP tag: fe71df57 To: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c SIP to address: sip:virtual number@kamailio SIP tag: as416d507c Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 1 INVITE Sequence Number: 1 Method: INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:virtual number@asterisk> Contact Binding: <sip:virtual number@asterisk> URI: <sip:virtual number@asterisk> SIP contact address: sip:virtual number@asterisk Content-Type: application/sdp Content-Length: 285 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 4184 4185 IN IP4 asterisk Owner Username: root Session ID: 4184 Session Version: 4185 Owner Network Type: IN Owner Address Type: IP4 Owner Address: asterisk Session Name (s): session Connection Information (c): IN IP4 asterisk Connection Network Type: IN Connection Address Type: IP4 Connection Address: asterisk Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 Media Description, name and address (m): audio 16620 RTP/AVP 3 0 8 101 Media Type: audio Media Port: 16620 Media Proto: RTP/AVP Media Format: GSM 06.10 Media Format: ITU-T G.711 PCMU Media Format: ITU-T G.711 PCMA Media Format: 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute Fieldname: rtpmap Media Format: 3 MIME Type: GSM Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Format: 0 MIME Type: PCMU Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute Fieldname: rtpmap Media Format: 8 MIME Type: PCMA Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Format: 101 MIME Type: telephone-event Media Attribute (a): fmtp:101 0-16 Media Attribute Fieldname: fmtp Media Format: 101 [telephone-event] Media format specific parameters: 0-16 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - Media Attribute (a): ptime:20 Media Attribute Fieldname: ptime Media Attribute Value: 20 Media Attribute (a): sendrecv
No. Time Source Destination Protocol Info 17 3.579216 kamailio sip subscriber A SIP/SDP Status: 183 Session Progress, with session description
Internet Protocol, Src: kamailio (kamailio), Dst: sip subscriber A (sip subscriber A) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 183 Session Progress Status-Code: 183 [Resent Packet: False] Message Header Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber A;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060 Transport: UDP Sent-by Address: sip subscriber A Sent-by port: 5060 Received: sip subscriber A Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z- RPort: 5060 Record-Route: sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887 From: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@kamailio SIP tag: fe71df57 To: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c SIP to address: sip:virtual number@kamailio SIP tag: as416d507c Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 1 INVITE Sequence Number: 1 Method: INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:virtual number@asterisk> Contact Binding: <sip:virtual number@asterisk> URI: <sip:virtual number@asterisk> SIP contact address: sip:virtual number@asterisk Content-Type: application/sdp Content-Length: 285 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 4184 4184 IN IP4 asterisk Owner Username: root Session ID: 4184 Session Version: 4184 Owner Network Type: IN Owner Address Type: IP4 Owner Address: asterisk Session Name (s): session Connection Information (c): IN IP4 asterisk Connection Network Type: IN Connection Address Type: IP4 Connection Address: asterisk Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 Media Description, name and address (m): audio 16620 RTP/AVP 3 0 8 101 Media Type: audio Media Port: 16620 Media Proto: RTP/AVP Media Format: GSM 06.10 Media Format: ITU-T G.711 PCMU Media Format: ITU-T G.711 PCMA Media Format: 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute Fieldname: rtpmap Media Format: 3 MIME Type: GSM Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Format: 0 MIME Type: PCMU Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute Fieldname: rtpmap Media Format: 8 MIME Type: PCMA Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Format: 101 MIME Type: telephone-event Media Attribute (a): fmtp:101 0-16 Media Attribute Fieldname: fmtp Media Format: 101 [telephone-event] Media format specific parameters: 0-16 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - Media Attribute (a): ptime:20 Media Attribute Fieldname: ptime Media Attribute Value: 20 Media Attribute (a): sendrecv
No. Time Source Destination Protocol Info 18 3.580413 kamailio sip subscriber A SIP/SDP Status: 200 OK, with session description
Internet Protocol, Src: kamailio (kamailio), Dst: sip subscriber A (sip subscriber A) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber A;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060 Transport: UDP Sent-by Address: sip subscriber A Sent-by port: 5060 Received: sip subscriber A Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z- RPort: 5060 Record-Route: sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887 From: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@kamailio SIP tag: fe71df57 To: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c SIP to address: sip:virtual number@kamailio SIP tag: as416d507c Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 1 INVITE Sequence Number: 1 Method: INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:virtual number@asterisk> Contact Binding: <sip:virtual number@asterisk> URI: <sip:virtual number@asterisk> SIP contact address: sip:virtual number@asterisk Content-Type: application/sdp Content-Length: 285 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 4184 4185 IN IP4 asterisk Owner Username: root Session ID: 4184 Session Version: 4185 Owner Network Type: IN Owner Address Type: IP4 Owner Address: asterisk Session Name (s): session Connection Information (c): IN IP4 asterisk Connection Network Type: IN Connection Address Type: IP4 Connection Address: asterisk Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 Media Description, name and address (m): audio 16620 RTP/AVP 3 0 8 101 Media Type: audio Media Port: 16620 Media Proto: RTP/AVP Media Format: GSM 06.10 Media Format: ITU-T G.711 PCMU Media Format: ITU-T G.711 PCMA Media Format: 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute Fieldname: rtpmap Media Format: 3 MIME Type: GSM Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Format: 0 MIME Type: PCMU Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute Fieldname: rtpmap Media Format: 8 MIME Type: PCMA Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Format: 101 MIME Type: telephone-event Media Attribute (a): fmtp:101 0-16 Media Attribute Fieldname: fmtp Media Format: 101 [telephone-event] Media format specific parameters: 0-16 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - Media Attribute (a): ptime:20 Media Attribute Fieldname: ptime Media Attribute Value: 20 Media Attribute (a): sendrecv
No. Time Source Destination Protocol Info 19 3.811307 sip subscriber A kamailio SIP Request: ACK sip:virtual number@asterisk
Internet Protocol, Src: sip subscriber A (sip subscriber A), Dst: kamailio (kamailio) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:virtual number@asterisk SIP/2.0 Method: ACK [Resent Packet: False] Message Header Via: SIP/2.0/UDP sip subscriber A:5060;branch=z9hG4bK-d8754z-5463cc5518cdcc67-1---d8754z-;rport Transport: UDP Sent-by Address: sip subscriber A Sent-by port: 5060 Branch: z9hG4bK-d8754z-5463cc5518cdcc67-1---d8754z- RPort: rport Max-Forwards: 70 Route: sip:kamailio;lr;ftag=fe71df57;did=9f7.5f04d887 Contact: <sip:sip subscriber A@sip subscriber A:5060> Contact Binding: <sip:sip subscriber A@sip subscriber A:5060> URI: <sip:sip subscriber A@sip subscriber A:5060> SIP contact address: sip:sip subscriber A@sip subscriber A:5060 To: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c SIP to address: sip:virtual number@kamailio SIP tag: as416d507c From: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@kamailio SIP tag: fe71df57 Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 1 ACK Sequence Number: 1 Method: ACK User-Agent: Zoiper rev.448 Content-Length: 0
No. Time Source Destination Protocol Info 20 3.812537 kamailio asterisk SIP Request: ACK sip:virtual number@asterisk
Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:virtual number@asterisk SIP/2.0 Method: ACK [Resent Packet: False] Message Header Via: SIP/2.0/UDP kamailio;branch=z9hG4bKce3c.92b7a847.2 Transport: UDP Sent-by Address: kamailio Branch: z9hG4bKce3c.92b7a847.2 Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber A;branch=z9hG4bK-d8754z-5463cc5518cdcc67-1---d8754z-;rport=5060 Transport: UDP Sent-by Address: sip subscriber A Sent-by port: 5060 Received: sip subscriber A Branch: z9hG4bK-d8754z-5463cc5518cdcc67-1---d8754z- RPort: 5060 Max-Forwards: 69 Contact: <sip:sip subscriber A@sip subscriber A:5060> Contact Binding: <sip:sip subscriber A@sip subscriber A:5060> URI: <sip:sip subscriber A@sip subscriber A:5060> SIP contact address: sip:sip subscriber A@sip subscriber A:5060 To: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c SIP to address: sip:virtual number@kamailio SIP tag: as416d507c From: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@kamailio SIP tag: fe71df57 Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 1 ACK Sequence Number: 1 Method: ACK User-Agent: Zoiper rev.448 Content-Length: 0
No. Time Source Destination Protocol Info 42 11.624423 asterisk kamailio SIP/SDP Request: INVITE sip:virtual number@kamailio, with session description
Internet Protocol, Src: asterisk (asterisk), Dst: kamailio (kamailio) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:virtual number@kamailio SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK384c143a;rport Transport: UDP Sent-by Address: asterisk Sent-by port: 5060 Branch: z9hG4bK384c143a RPort: rport From: "sip subscriber A" <sip:sip subscriber A@asterisk>;tag=as59e5678f SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@asterisk SIP tag: as59e5678f To: <sip:virtual number@kamailio> SIP to address: sip:virtual number@kamailio Contact: <sip:sip subscriber A@asterisk> Contact Binding: <sip:sip subscriber A@asterisk> URI: <sip:sip subscriber A@asterisk> SIP contact address: sip:sip subscriber A@asterisk Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk CSeq: 102 INVITE Sequence Number: 102 Method: INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 09 Mar 2010 17:30:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 285 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 4184 4184 IN IP4 asterisk Owner Username: root Session ID: 4184 Session Version: 4184 Owner Network Type: IN Owner Address Type: IP4 Owner Address: asterisk Session Name (s): session Connection Information (c): IN IP4 asterisk Connection Network Type: IN Connection Address Type: IP4 Connection Address: asterisk Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 Media Description, name and address (m): audio 13698 RTP/AVP 3 0 8 101 Media Type: audio Media Port: 13698 Media Proto: RTP/AVP Media Format: GSM 06.10 Media Format: ITU-T G.711 PCMU Media Format: ITU-T G.711 PCMA Media Format: 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute Fieldname: rtpmap Media Format: 3 MIME Type: GSM Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Format: 0 MIME Type: PCMU Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute Fieldname: rtpmap Media Format: 8 MIME Type: PCMA Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Format: 101 MIME Type: telephone-event Media Attribute (a): fmtp:101 0-16 Media Attribute Fieldname: fmtp Media Format: 101 [telephone-event] Media format specific parameters: 0-16 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - Media Attribute (a): ptime:20 Media Attribute Fieldname: ptime Media Attribute Value: 20 Media Attribute (a): sendrecv
No. Time Source Destination Protocol Info 43 11.625328 kamailio asterisk SIP Status: 100 Giving a try
Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Giving a try Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK384c143a;rport=5060 Transport: UDP Sent-by Address: asterisk Sent-by port: 5060 Branch: z9hG4bK384c143a RPort: 5060 From: "sip subscriber A" <sip:sip subscriber A@asterisk>;tag=as59e5678f SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@asterisk SIP tag: as59e5678f To: <sip:virtual number@kamailio> SIP to address: sip:virtual number@kamailio Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk CSeq: 102 INVITE Sequence Number: 102 Method: INVITE Server: Kamailio (1.5.3-notls (x86_64/linux)) Content-Length: 0
No. Time Source Destination Protocol Info 44 11.625386 kamailio PSTN SIP/SDP Request: INVITE sip:6937630910@PSTN, with session description
Internet Protocol, Src: kamailio (kamailio), Dst: PSTN (PSTN) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:6937630910@PSTN SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Record-Route: sip:kamailio;lr=on;ftag=as59e5678f;did=631.33197276 Via: SIP/2.0/UDP kamailio;branch=z9hG4bK2b67.85f739f3.0 Transport: UDP Sent-by Address: kamailio Branch: z9hG4bK2b67.85f739f3.0 Via: SIP/2.0/UDP asterisk:5060;received=asterisk;branch=z9hG4bK384c143a;rport=5060 Transport: UDP Sent-by Address: asterisk Sent-by port: 5060 Received: asterisk Branch: z9hG4bK384c143a RPort: 5060 From: "sip subscriber A" <sip:sip subscriber A@asterisk>;tag=as59e5678f SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@asterisk SIP tag: as59e5678f To: <sip:virtual number@kamailio> SIP to address: sip:virtual number@kamailio Contact: <sip:sip subscriber A@asterisk> Contact Binding: <sip:sip subscriber A@asterisk> URI: <sip:sip subscriber A@asterisk> SIP contact address: sip:sip subscriber A@asterisk Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk CSeq: 102 INVITE Sequence Number: 102 Method: INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Tue, 09 Mar 2010 17:30:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 285 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 4184 4184 IN IP4 asterisk Owner Username: root Session ID: 4184 Session Version: 4184 Owner Network Type: IN Owner Address Type: IP4 Owner Address: asterisk Session Name (s): session Connection Information (c): IN IP4 asterisk Connection Network Type: IN Connection Address Type: IP4 Connection Address: asterisk Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 Media Description, name and address (m): audio 13698 RTP/AVP 3 0 8 101 Media Type: audio Media Port: 13698 Media Proto: RTP/AVP Media Format: GSM 06.10 Media Format: ITU-T G.711 PCMU Media Format: ITU-T G.711 PCMA Media Format: 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute Fieldname: rtpmap Media Format: 3 MIME Type: GSM Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Format: 0 MIME Type: PCMU Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute Fieldname: rtpmap Media Format: 8 MIME Type: PCMA Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Format: 101 MIME Type: telephone-event Media Attribute (a): fmtp:101 0-16 Media Attribute Fieldname: fmtp Media Format: 101 [telephone-event] Media format specific parameters: 0-16 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - Media Attribute (a): ptime:20 Media Attribute Fieldname: ptime Media Attribute Value: 20 Media Attribute (a): sendrecv
No. Time Source Destination Protocol Info 45 11.626223 PSTN kamailio SIP Status: 100 Trying
Internet Protocol, Src: PSTN (PSTN), Dst: kamailio (kamailio) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP kamailio;branch=z9hG4bK2b67.85f739f3.0 Transport: UDP Sent-by Address: kamailio Branch: z9hG4bK2b67.85f739f3.0 Via: SIP/2.0/UDP asterisk:5060;received=asterisk;branch=z9hG4bK384c143a;rport=5060 Transport: UDP Sent-by Address: asterisk Sent-by port: 5060 Received: asterisk Branch: z9hG4bK384c143a RPort: 5060 From: "sip subscriber A" <sip:sip subscriber A@asterisk>;tag=as59e5678f SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@asterisk SIP tag: as59e5678f To: <sip:virtual number@kamailio> SIP to address: sip:virtual number@kamailio Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk CSeq: 102 INVITE Sequence Number: 102 Method: INVITE Content-Length: 0
No. Time Source Destination Protocol Info 46 11.626916 PSTN kamailio SIP Status: 503 Service Unavailable
Internet Protocol, Src: PSTN (PSTN), Dst: kamailio (kamailio) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 503 Service Unavailable Status-Code: 503 [Resent Packet: False] Message Header Via: SIP/2.0/UDP kamailio;branch=z9hG4bK2b67.85f739f3.0 Transport: UDP Sent-by Address: kamailio Branch: z9hG4bK2b67.85f739f3.0 Via: SIP/2.0/UDP asterisk:5060;received=asterisk;branch=z9hG4bK384c143a;rport=5060 Transport: UDP Sent-by Address: asterisk Sent-by port: 5060 Received: asterisk Branch: z9hG4bK384c143a RPort: 5060 Record-Route: sip:kamailio;lr=on;ftag=as59e5678f;did=631.33197276 To: <sip:virtual number@kamailio>;tag=3477144611-878554 SIP to address: sip:virtual number@kamailio SIP tag: 3477144611-878554 From: "sip subscriber A" <sip:sip subscriber A@asterisk>;tag=as59e5678f SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@asterisk SIP tag: as59e5678f Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk CSeq: 102 INVITE Sequence Number: 102 Method: INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK Contact: sip:6937630910@PSTN:5060 Contact Binding: sip:6937630910@PSTN:5060 URI: sip:6937630910@PSTN:5060 SIP contact address: sip:6937630910@PSTN:5060 Call-Info: sip:PSTN;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0
No. Time Source Destination Protocol Info 47 11.627296 kamailio PSTN SIP Request: ACK sip:6937630910@PSTN
Internet Protocol, Src: kamailio (kamailio), Dst: PSTN (PSTN) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:6937630910@PSTN SIP/2.0 Method: ACK [Resent Packet: False] Message Header Via: SIP/2.0/UDP kamailio;branch=z9hG4bK2b67.85f739f3.0 Transport: UDP Sent-by Address: kamailio Branch: z9hG4bK2b67.85f739f3.0 From: "sip subscriber A" <sip:sip subscriber A@asterisk>;tag=as59e5678f SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@asterisk SIP tag: as59e5678f Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk To: <sip:virtual number@kamailio>;tag=3477144611-878554 SIP to address: sip:virtual number@kamailio SIP tag: 3477144611-878554 CSeq: 102 ACK Sequence Number: 102 Method: ACK Max-Forwards: 70 User-Agent: Kamailio (1.5.3-notls (x86_64/linux)) Content-Length: 0
No. Time Source Destination Protocol Info 48 11.627838 kamailio asterisk SIP Status: 444 No more tries for you!
Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 444 No more tries for you! Status-Code: 444 [Resent Packet: False] Message Header Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK384c143a;rport=5060 Transport: UDP Sent-by Address: asterisk Sent-by port: 5060 Branch: z9hG4bK384c143a RPort: 5060 From: "sip subscriber A" <sip:sip subscriber A@asterisk>;tag=as59e5678f SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@asterisk SIP tag: as59e5678f To: <sip:virtual number@kamailio>;tag=0076135696eb5d2fff122699f03f5620-de63 SIP to address: sip:virtual number@kamailio SIP tag: 0076135696eb5d2fff122699f03f5620-de63 Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk CSeq: 102 INVITE Sequence Number: 102 Method: INVITE Server: Kamailio (1.5.3-notls (x86_64/linux)) Content-Length: 0
No. Time Source Destination Protocol Info 49 11.628045 asterisk kamailio SIP Request: ACK sip:virtual number@kamailio
Internet Protocol, Src: asterisk (asterisk), Dst: kamailio (kamailio) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:virtual number@kamailio SIP/2.0 Method: ACK [Resent Packet: False] Message Header Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK384c143a;rport Transport: UDP Sent-by Address: asterisk Sent-by port: 5060 Branch: z9hG4bK384c143a RPort: rport From: "sip subscriber A" <sip:sip subscriber A@asterisk>;tag=as59e5678f SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@asterisk SIP tag: as59e5678f To: <sip:virtual number@kamailio>;tag=0076135696eb5d2fff122699f03f5620-de63 SIP to address: sip:virtual number@kamailio SIP tag: 0076135696eb5d2fff122699f03f5620-de63 Contact: <sip:sip subscriber A@asterisk> Contact Binding: <sip:sip subscriber A@asterisk> URI: <sip:sip subscriber A@asterisk> SIP contact address: sip:sip subscriber A@asterisk Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk CSeq: 102 ACK Sequence Number: 102 Method: ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0
No. Time Source Destination Protocol Info 50 11.629887 asterisk kamailio SIP Request: BYE sip:sip subscriber A@sip subscriber A:5060
Internet Protocol, Src: asterisk (asterisk), Dst: kamailio (kamailio) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: BYE sip:sip subscriber A@sip subscriber A:5060 SIP/2.0 Method: BYE [Resent Packet: False] Message Header Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK514b2ee7;rport Transport: UDP Sent-by Address: asterisk Sent-by port: 5060 Branch: z9hG4bK514b2ee7 RPort: rport Route: sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887 From: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c SIP from address: sip:virtual number@kamailio SIP tag: as416d507c To: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP to address: sip:sip subscriber A@kamailio SIP tag: fe71df57 Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 102 BYE Sequence Number: 102 Method: BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0
No. Time Source Destination Protocol Info 51 11.630426 kamailio sip subscriber A SIP Request: BYE sip:sip subscriber A@sip subscriber A:5060
Internet Protocol, Src: kamailio (kamailio), Dst: sip subscriber A (sip subscriber A) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: BYE sip:sip subscriber A@sip subscriber A:5060 SIP/2.0 Method: BYE [Resent Packet: False] Message Header Via: SIP/2.0/UDP kamailio;branch=z9hG4bK482e.f4934753.0 Transport: UDP Sent-by Address: kamailio Branch: z9hG4bK482e.f4934753.0 Via: SIP/2.0/UDP asterisk:5060;received=asterisk;branch=z9hG4bK514b2ee7;rport=5060 Transport: UDP Sent-by Address: asterisk Sent-by port: 5060 Received: asterisk Branch: z9hG4bK514b2ee7 RPort: 5060 From: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c SIP from address: sip:virtual number@kamailio SIP tag: as416d507c To: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP to address: sip:sip subscriber A@kamailio SIP tag: fe71df57 Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 102 BYE Sequence Number: 102 Method: BYE User-Agent: Asterisk PBX Max-Forwards: 69 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0
No. Time Source Destination Protocol Info 52 11.712546 sip subscriber A kamailio SIP Status: 200 OK
Internet Protocol, Src: sip subscriber A (sip subscriber A), Dst: kamailio (kamailio) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP kamailio;branch=z9hG4bK482e.f4934753.0 Transport: UDP Sent-by Address: kamailio Branch: z9hG4bK482e.f4934753.0 Via: SIP/2.0/UDP asterisk:5060;received=asterisk;branch=z9hG4bK514b2ee7;rport=5060 Transport: UDP Sent-by Address: asterisk Sent-by port: 5060 Received: asterisk Branch: z9hG4bK514b2ee7 RPort: 5060 Contact: <sip:sip subscriber A@sip subscriber A:5060> Contact Binding: <sip:sip subscriber A@sip subscriber A:5060> URI: <sip:sip subscriber A@sip subscriber A:5060> SIP contact address: sip:sip subscriber A@sip subscriber A:5060 To: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP to address: sip:sip subscriber A@kamailio SIP tag: fe71df57 From: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c SIP from address: sip:virtual number@kamailio SIP tag: as416d507c Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 102 BYE Sequence Number: 102 Method: BYE User-Agent: Zoiper rev.448 Content-Length: 0
No. Time Source Destination Protocol Info 53 11.712663 kamailio asterisk SIP Status: 200 OK
Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP asterisk:5060;received=asterisk;branch=z9hG4bK514b2ee7;rport=5060 Transport: UDP Sent-by Address: asterisk Sent-by port: 5060 Received: asterisk Branch: z9hG4bK514b2ee7 RPort: 5060 Contact: <sip:sip subscriber A@sip subscriber A:5060> Contact Binding: <sip:sip subscriber A@sip subscriber A:5060> URI: <sip:sip subscriber A@sip subscriber A:5060> SIP contact address: sip:sip subscriber A@sip subscriber A:5060 To: "sip subscriber A"<sip:sip subscriber A@kamailio>;transport=UDP;tag=fe71df57 SIP Display info: "sip subscriber A" SIP to address: sip:sip subscriber A@kamailio SIP tag: fe71df57 From: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c SIP from address: sip:virtual number@kamailio SIP tag: as416d507c Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA. CSeq: 102 BYE Sequence Number: 102 Method: BYE User-Agent: Zoiper rev.448 Content-Length: 0
No. Time Source Destination Protocol Info 62 12.102418 kamailio asterisk SIP Status: 444 No more tries for you!
Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 444 No more tries for you! Status-Code: 444 [Resent Packet: True] [Suspected resend of frame: 48] Message Header Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK384c143a;rport=5060 Transport: UDP Sent-by Address: asterisk Sent-by port: 5060 Branch: z9hG4bK384c143a RPort: 5060 From: "sip subscriber A" <sip:sip subscriber A@asterisk>;tag=as59e5678f SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@asterisk SIP tag: as59e5678f To: <sip:virtual number@kamailio>;tag=0076135696eb5d2fff122699f03f5620-de63 SIP to address: sip:virtual number@kamailio SIP tag: 0076135696eb5d2fff122699f03f5620-de63 Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk CSeq: 102 INVITE Sequence Number: 102 Method: INVITE Server: Kamailio (1.5.3-notls (x86_64/linux)) Content-Length: 0
No. Time Source Destination Protocol Info 73 13.102386 kamailio asterisk SIP Status: 444 No more tries for you!
Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 444 No more tries for you! Status-Code: 444 [Resent Packet: True] [Suspected resend of frame: 48] Message Header Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK384c143a;rport=5060 Transport: UDP Sent-by Address: asterisk Sent-by port: 5060 Branch: z9hG4bK384c143a RPort: 5060 From: "sip subscriber A" <sip:sip subscriber A@asterisk>;tag=as59e5678f SIP Display info: "sip subscriber A" SIP from address: sip:sip subscriber A@asterisk SIP tag: as59e5678f To: <sip:virtual number@kamailio>;tag=0076135696eb5d2fff122699f03f5620-de63 SIP to address: sip:virtual number@kamailio SIP tag: 0076135696eb5d2fff122699f03f5620-de63 Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk CSeq: 102 INVITE Sequence Number: 102 Method: INVITE Server: Kamailio (1.5.3-notls (x86_64/linux)) Content-Length: 0
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:
- A Call is placed by a sip subscriber "A"
- kamailio forwards the call to the asterisk server
- 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}); # Dialplangroup 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 numbertranslation 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 differentprefixes $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 finalRURI 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
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Daniel-Constantin Mierla -
marius zbihlei -
Panagiotis Skoulikaritis