July 2012 - sr-users - kamailio.org

Problem with t_relay() and VPN tunnel
by Carlos Ruiz Diaz 24 Jun '17

24 Jun '17

Hi, I have two SIP servers, *A* and *B*, connected each other though a OPENVPN tunnel. The server *B* needs to t_relay() every SIP message containing the method MESSAGE to the server *A* but these messages never reach destination. I have tested the tunnel connectivity and works fine. I wrote a Perl script (located in *B*) that sends SIP MESSAGES to Kamailio (located in *A*) trying to figure out what is happening but these messages are received by *A* and processed correctly but when *B* does the same from Kamailio, it is never received. Here is the route part of kamailio.cfg in *B*: Observation: ($rU == "1004") result is *true* * if(is_method("MESSAGE")) { if($rU == "1004") { xlog("L_INFO","En 1004"); rewritehost("10.8.0.1"); if (!t_relay()) xlog("L_INFO","MIO Error en t_relay"); t_reply("200", "Ok"); xlog("L_INFO","MIO despues rewrite"); exit; } .....* The perl script that WORKS: *$msg = 'MESSAGE sip:1004@192.168.1.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2 From: "2002" <sip:2002@192.168.1.2 <sip%3A2002(a)192.168.1.2>>;tag=1837944796 To: <sip:1004@192.168.1.2 <sip%3A1004(a)192.168.1.2>> Call-ID: 19722852989(a)192.168.1.2 CSeq: 15773 MESSAGE Contact: <sip:2002@**PublicIP-protected**:5060> Max-Forwards: 29 User-Agent: DBL Content-Type: text/plain Content-Length: 34 +595981[protected] hello from kamailio '; use IO::Socket; my $sock = IO::Socket::INET->new( Proto=>'udp', PeerHost=>'192.168.2.102', PeerPort=>'5060'); print "Sending msg $msg\n"; $sock->send($msg) or die "error sending $!\n"; *Please help! Thanks in advance. Carlos.

3 3

[Kamailio-Users] HA+DRBD setup
by Siddhardha Garige 09 Jul '15

09 Jul '15

Hello all, Just wondering if anyone know any tutorial on setting up HA+DRBD solution for kamailio. Especially creating partitions, DRBD devices and mount points. Thanks in advance, -Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

11 24

Re: [Kamailio-Users] /usr/local/sbin/kamdbctl create error
by vivi 01 Nov '14

01 Nov '14

Hi all.. I have solve the problem . Cheers, vivi From: vivi [mailto:vivi.hilton@gmail.com] Sent: Saturday, June 06, 2009 11:29 PM To: 'users(a)lists.kamailio.org' Subject: /usr/local/sbin/kamdbctl create error Hi all I have specify the wanted db type (DBENGINE=MYSQL) in the /usr/local/etc/kamailio/kamctlrc then I using "/usr/local/sbin/kamdbctl create" to create MySQL database, but I got this error: ERROR: database engine not specified, please setup one in the config script root@acer:/usr/local/src/kamailio-1.5.0/sip-server# vim /usr/local/etc/kamailio/kamctlrc root@acer:/usr/local/src/kamailio-1.5.0/sip-server# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR: Creating core database and grant privileges failed! Cheers, vivi

2 1

kamailioi is not relaying BYE message to UAC
by Arif Hossain 06 Dec '13

06 Dec '13

Hi, We have the following network architecture : UAC1------------------------->kamailio-------------------->VoipSwitch----->PSTN---------->Phone1 (Sip Client) Now UAC1 calls Phone1 and everything is ok. If UAC1 hangs up session is terminated cleanly. But if Phone1 hangs up the BYE message which comes to kamailio and goes back to VoipSwitch instead of relayed to UAC1 . So The session becomes a zombie one, And UAC1 unfortunately gets billed for a session which should be terminated. Following is the Call flow as seen from VoipSwitch : | <kamilio IP> | | | | <VoipSwitch IP> | |134.856 | INVITE SDP | |(7890) ------------------> (5060) | |134.858 | 407 Proxy Authentication Required | |(7890) <------------------ (5060) | |134.902 | ACK | | |(7890) ------------------> (5060) | |135.408 | INVITE SDP | |(7890) ------------------> (5060) | |135.414 | 100 Trying| | |(7890) <------------------ (5060) | |140.121 | 183 Session Progress SDP | |(7890) <------------------ (5060) | |140.184 | RTP (g729) | | |(61868) <------------------ (5136) | |141.295 | RTP (g729) | | |(61868) ------------------> (5136) | |153.701 | 200 OK SDP | |(7890) <------------------ (5060) | |153.713 | RTP (g729) | | |(61868) ------------------> (5136) | |154.126 | ACK | | |(7890) ------------------> (5060) | |159.988 | BYE | | |(7890) <------------------ (5060) | |160.031 | BYE | | |(7890) ------------------> (5060) | |160.478 | BYE | | |(7890) ------------------> (5060) | |161.412 | BYE | | |(7890) ------------------> (5060) | |181.952 | BYE | | |(7890) ------------------> (5060) | |185.687 | BYE | | |(7890) ------------------> (5060) | |188.018 | 408 Request Timeout | |(7890) ------------------> (5060) | Sip Traces : kamailio------>VoipSwitch I'm posting only the offending BYE msg instead of full trace , because of the mail will become difficult to read . If more traces needed, i can post it. The following BYE message is sent by VoipSwitch: BYE sip:ipphone@205.164.40.74 SIP/2.0 Route: <sip:108.166.195.189:7890;lr=on;nat=yes> CSeq: 2 BYE Via: SIP/2.0/UDP 205.164.40.74:5060 From: sip:008801673345531@205.164.40.74;tag=100528120745985872655137 Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS To: "123456" <sip:ipphone@205.164.40.74>;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG Content-Length: 0 Kamailio instead of relaying the message, sends a BYE message towards VoipSwitch: BYE sip:ipphone@205.164.40.74 SIP/2.0 Max-Forwards: 10 CSeq: 2 BYE Via: SIP/2.0/UDP 108.166.195.189:7890;branch=z9hG4bK4b2b.5d893e95.0 Via: SIP/2.0/UDP 205.164.40.74:5060;rport=5060 From: sip:008801673345531@205.164.40.74;tag=100528120745985872655137 Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS To: "123456" <sip:ipphone@205.164.40.74>;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG Content-Length: 0 When the first BYE message comes from VoipSwitch , kamailio does the following : May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: <core> [receive.c:289]: receive_msg: cleaning up May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: <core> [parser/sdp/sdp.c:751]: _sdp = 0x831bf10 May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: <core> [parser/sdp/sdp.c:753]: sdp = 0x83043dc May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: <core> [parser/sdp/sdp.c:755]: session = 0x8304504 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:630]: SIP Request: May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:632]: method: <BYE> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:634]: uri: <sip:ipphone@205.164.40.74> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <1> <BYE> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bk19052612230719933454843>; state=16 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [receive.c:145]: After parse_msg... May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [receive.c:186]: preparing to run routing scripts... May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=arILprdVR1srJ76HHlt4BEc3XsyaWcZm May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_field: <To> [76]; uri=[sip:ipphone@205.164.40.74] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:189]: DEBUG: to body ["ipphone" <sip:ipphone@205.164.40.74>] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: maxfwd [mf_funcs.c:66]: max_forwards header not found! May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=1905251223419334290936029 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: siputils [checks.c:76]: totag found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:85]: is_preloaded: No May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==15 && [205.164.40.74] == [108.166.195.189] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 7890 matches port 5060 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==15 && [205.164.40.74] == [108.166.195.189] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [forward.c:448]: check_self: host != me May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 15==15 && [108.166.195.189] == [108.166.195.189] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 7890 matches port 7890 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:792]: Topmost route URI: 'sip:108.166.195.189:7890;lr=on;nat=yes' is me May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:257]: No next Route HF found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:811]: No next URI found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:983]: params are <;lr=on;nat=yes> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: siputils [checks.c:76]: totag found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=2501 , global msg id=2500 , T on entrance=0xffffffff May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=29177, isACK=0 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:564]: DEBUG: proceeding to pre-RFC3261 transaction matching May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb61626a4, callback type 1, id 0 entered May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [msg_translator.c:204]: check_via_address(205.164.40.74, 205.164.40.74, 0) May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) May 20 02:26:04 VOS20-108 last message repeated 5 times May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [receive.c:289]: receive_msg: cleaning up May 20 02:26:22 VOS20-108 /usr/local/sbin/kamailio[16445]: DEBUG: <core> [udp_server.c:486]: udp_rcv_loop: probing packet received from 180.234.62.230 38722 May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm [t_reply.c:1134]: ->>>>>>>>> T_code=0, new_code=408 May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm [t_reply.c:1636]: DEBUG: relay_reply: branch=0, save=0, relay=0 May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: <core> [msg_translator.c:204]: check_via_address(205.164.40.74, 205.164.40.74, 0) May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: <core> [mem/shm_mem.c:105]: WARNING:vqm_resize: resize(0) called May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm [t_hooks.c:288]: DBG: trans=0xb61626a4, callback type 128, id 0 entered May 20 02:26:54 VOS20-108 /usr/local/sbin/kamailio[16451]: WARNING: <core> [timer.c:450]: WARNING: our timer runs faster then real-time (300000 ms / 4800 ticks our time .-> 299923 ms / 4798 ticks real time) OT: How do you guys maintain 80 column mails? i do not use a mail client, use gmail mailbox. -- -aft

5 6

EOF Error in pass_fd
by Ricardo Martinez 20 Nov '13

20 Nov '13

Hello List. The last weekend our kamailio process crashed with this error : Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10661]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 29 Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: ALERT: <core> [main.c:751]: child process 10601 exited by a signal 11 Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: ALERT: <core> [main.c:754]: core was generated Jun 24 06:55:08 pxh /usr/local/sbin/kamailio[10542]: INFO: <core> [main.c:766]: INFO: terminating due to SIGCHLD Can someone tell me what seems to be the problem here? We’re using : version: kamailio 3.2.2 (x86_64/linux) 98ba92-dirty flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 98ba92 -dirty compiled on 12:45:36 Feb 2 2012 with gcc 4.4.6 And this is the bt full from gdb Core was generated by `/usr/local/sbin/kamailio -m 1024'. Program terminated with signal 11, Segmentation fault. #0 mi_profile_list (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_profile.c:789 789 if ( mi_print_dlg( rpl, ph->dlg, 0)!=0 ) Missing separate debuginfos, use: debuginfo-install filesystem-2.4.30-2.1.el6.x86_64 glibc-2.12-1.25.el6_1.3.x86_64 keyutils-libs-1.4-1.el6.x86_64 krb5-libs-1.9-9.el6_1.2.x86_64 libcom_err-1.41.12-7.el6.x86_64 libselinux-2.0.94-5.el6.x86_64 mysql-libs-5.1.52-1.el6_0.1.x86_64 nss-softokn-freebl-3.12.7-1.1.el6.x86_64 openssl-1.0.0-20.el6_2.1.x86_64 pcre-7.8-3.1.el6.x86_64 zlib-1.2.3-26.el6.x86_64 (gdb) bt full #0 mi_profile_list (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_profile.c:789 node = <value optimized out> rpl_tree = 0x20a6790 rpl = 0x20a67b0 profile = 0x7fe99a2dda00 ph = 0x0 profile_name = <value optimized out> value = <value optimized out> i = <value optimized out> #1 0x00007fe9dce287b1 in run_mi_cmd (fifo_stream=<value optimized out>) at ../../lib/kmi/mi.h:77 No locals. #2 mi_fifo_server (fifo_stream=<value optimized out>) at fifo_fnc.c:509 mi_cmd = 0x208f8f0 mi_rpl = <value optimized out> hdl = 0x0 line_len = 41 file_sep = 0x2039017 "" command = 0x7fe9dfdbdef9 "profile_list_dlgs" file = 0x7fe9dfd98d78 "/tmp/openser_receiver_27809" f = 0x2039018 reply_stream = 0x20b0980 #3 0x00007fe9dce2a640 in fifo_process (rank=<value optimized out>) at mi_fifo.c:247 fifo_stream = 0x209a0b0 #4 0x00007fe9dce2a9de in mi_child_init (rank=0) at mi_fifo.c:211 pid = <value optimized out> #5 0x00000000004dba41 in init_mod_child (m=0x7fe9dfc6ba10, rank=0) at sr_module.c:886 No locals. #6 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6bd08, rank=0) at sr_module.c:883 No locals. #7 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6c248, rank=0) at sr_module.c:883 No locals. #8 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6c748, rank=0) at sr_module.c:883 No locals. #9 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6d288, rank=0) at sr_module.c:883 No locals. #10 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6d928, rank=0) at sr_module.c:883 No locals. #11 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6dee0, rank=0) at sr_module.c:883 No locals. #12 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6e1f0, rank=0) at sr_module.c:883 No locals. #13 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6e5d0, rank=0) at sr_module.c:883 No locals. #14 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6e830, rank=0) at sr_module.c:883 No locals. #15 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6ebd0, rank=0) at sr_module.c:883 No locals. #16 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6f220, rank=0) at sr_module.c:883 No locals. #17 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6f648, rank=0) at sr_module.c:883 No locals. #18 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6fa60, rank=0) at sr_module.c:883 No locals. #19 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc6fe40, rank=0) at sr_module.c:883 No locals. #20 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc702c0, rank=0) at sr_module.c:883 No locals. #21 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc70ae8, rank=0) at sr_module.c:883 No locals. ---Type <return> to continue, or q <return> to quit--- #22 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc70e18, rank=0) at sr_module.c:883 No locals. #23 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc71100, rank=0) at sr_module.c:883 No locals. #24 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc716c0, rank=0) at sr_module.c:883 No locals. #25 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc71df0, rank=0) at sr_module.c:883 No locals. #26 0x00000000004db9c4 in init_mod_child (m=0x7fe9dfc723c0, rank=0) at sr_module.c:883 No locals. #27 0x000000000046077d in main_loop () at main.c:1661 i = <value optimized out> pid = <value optimized out> si = 0x0 si_desc = "udp receiver child=15 sock=64.76.154.35:5060", '\000' <repeats 20 times>, "\003", '\000' <repeats 15 times>, "\030\325\a\232\351\177\000\000\001\000\000\000\002", '\000' <repeats 11 times>, "\001\000\000\000\000\000\000\000\003\000\000\000\000\000\000\000)\000\000\000\000\000\000" #28 0x0000000000461f13 in main (argc=<value optimized out>, argv=0x7fff2f2b2848) at main.c:2475 cfg_stream = <value optimized out> c = <value optimized out> r = <value optimized out> tmp = 0x7fff2f2b3f76 "" tmp_len = 0 options = 0x5a3178 ":f:cm:M:dVhEb:l:L:n:vrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 1850310455 rfd = <value optimized out> debug_save = <value optimized out> debug_flag = <value optimized out> dont_fork_cnt = <value optimized out> n_lst = <value optimized out> p = <value optimized out> (gdb) Regards, Ricardo.-

4 7

[Kamailio-Users] Error in Running kamailio !!
by Dinesh gautam 25 Jul '13

25 Jul '13

Dear Sir, I am using kamailio 1.5.0, having installed and configured as given on the kamailio site <http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn> on my PC having Centos operating system. I have configured the config files kamailio.cfg, kamctlrc accordingly as given on site. when I start the kamailio using kamctl start command. it gives the error message like *[root@localhost ~]# kamctl start database engine 'MYSQL' loaded Control engine 'FIFO' loaded INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed *please* *help me out to solve this problem and suggest me where I am wrong. * *I have configured kamalio so many times it gives me the same kind of error message. * * - With Regards, Dinesh Kumar Gautam Noida, +919911838275

6 15

File kamailio_fifo is removed...
by Bruno Bresciani 12 Mar '13

12 Mar '13

Hi, Kamailio is running fine, but when I try run again by command line without killing the previous, the kamailio_fifo is removed and functions fifo in kamctl doen't work anymore. Why when I try running kamailio when already exist one, it remove de file kamailio_fifo? I can't find the part of source code that file kamailio_fifo is removed... Someone knows? Best Regards

2 2

Wrong ACK packet sent from Cisco gw
by Nawfel Oujdi 07 Mar '13

07 Mar '13

Hello! I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK. ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 Route: <sip:911111500@PBX:5060> To: <sip:911111500@sip_proxy>;tag=as7f388e3f Date: Mon, 17 Jan 2011 09:26:36 GMT Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw To work fine , the content of Route header should be in ACK header and viceversa. I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header. INVITE sip:911111500@sip_proxy:5060 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 To: <sip:911111500@sip_proxy> i m using c5300-is-mz.123-26.bin ios version. Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested. Thanks in advance. Nawfel Oujdi here is the result of ngrep: U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 6. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: OpenSIPS (1.6.3-notls (i386/linux)). Content-Length: 0. . U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 5. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Length: 0. . U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. -- ------------------------------ Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s) dirección(es) indicadas anteriormente; el carácter confidencial, personal e intransferible del mismo está protegido legalmente. Cualquier revelación, uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha recibido este mensaje por equivocación, notifíquelo inmediatamente a la persona que lo ha enviado y borre el mensaje original junto con sus ficheros anexos sin leerlo ni grabarlo, total o parcialmente.

2 1

Kamailio 1.5 and Asterisk 1.8 issue (Protocol Error, Unspecified)
by Rowie 24 Feb '13

24 Feb '13

Hi, We are having an issue where a phone (snom in particular) cannot make a call through Asterisk. It just hangup and does not allow the call to go through. I am including a a sip trace on this thread to show what is happening within the call. Please see below: Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 -- View this message in context: http://old.nabble.com/Kamailio-1.5-and-Asterisk-1.8-issue-%28Protocol-Error… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

4 3

Core fm_realloc function
by Bruno Bresciani 08 Jan '13

08 Jan '13

Hi all, I observed two at 0x08173bf0 in fm_realloc (qm=0xb5f7d000, p=0xb61f0bc0, size=0) at mem/f_malloc.c:536. I don't know exactly what may be caused this core because the debug of kamailio is disabled, but looking at backtrace (gdb) and source code I could notice a relationship with TLS connection establishment (handshake). Other important fact is that core was generated by the kamalio itself with signal 6 (abort) because it can't find the pointer to memory realloc. I'd like to know if someone already observed this issue and if my analyse above is correct? Bellow is backtrace full with gdb: Core was generated by `/home2/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid'. Program terminated with signal 6, Aborted. #0 0x00f30402 in __kernel_vsyscall () (gdb) (gdb) (gdb) (gdb) (gdb) bt full #0 0x00f30402 in __kernel_vsyscall () No symbol table info available. #1 0x009fec10 in raise () from /lib/libc.so.6 No symbol table info available. #2 0x00a00521 in abort () from /lib/libc.so.6 No symbol table info available. #3 0x08173bf0 in fm_realloc (qm=0xb5f7d000, p=0xb61f0bc0, size=0) at mem/f_malloc.c:536 f = (struct fm_frag *) 0xb61f0bb8 pf = <value optimized out> orig_size = 18 L=ÃbÂ¬Ã2n = (struct fm_frag *) 0xb61f0bd2 ptr = <value optimized out> hash = 23548 #4 0x0051a87f in ser_realloc (ptr=0xb61f0bc0, size=32) at ../../mem/shm_mem.h:266 No locals. #5 0x00460295 in OPENSSL_cleanse () from /lib/libcrypto.so.6 No symbol table info available. #6 0x004609f7 in CRYPTO_realloc () from /lib/libcrypto.so.6 No symbol table info available. #7 0x003efb22 in sk_insert () from /lib/libcrypto.so.6 No symbol table info available. #8 0x003efbaa in sk_push () from /lib/libcrypto.so.6 No symbol table info available. #9 0x007c2ac3 in ssl_bytes_to_cipher_list () from /lib/libssl.so.6 No symbol table info available. #10 0x007aa9f3 in ssl3_get_client_hello () from /lib/libssl.so.6 No symbol table info available. #11 0x007ab815 in ssl3_accept () from /lib/libssl.so.6 No symbol table info available. #12 0x007c1c3a in SSL_accept () from /lib/libssl.so.6 No symbol table info available. #13 0x005278e3 in tls_accept (c=0xb621af88, error=0xbf91dfc8) at tls_server.c:346 ret = <value optimized out> ssl = (SSL *) 0xb61c1878 cert = <value optimized out> tls_c = (struct tls_extra_data *) 0xb6208e58 tls_log = <value optimized out> #14 0x0052da8e in tls_read_f (c=0xb621af88, flags=0xbf91e200) at tls_server.c:1028 r = (struct tcp_req *) 0xb621affc bytes_free = 4095 bytes_read = 70 read_size = 4095 ssl_error = 0 ssl_read = 0 ssl = (SSL *) 0xb61c1878 rd_buf = "\026\003\001\000A\001\000\000=\003\001OÃ¿\vÂ¡ywP\fkÃDÃ 0/\000\a\000\005\001\000ÃÃÃÃ¯000\000\026\0009\0008\0005\000\026\000\023\000\n\0003\0002\00 202i<Ã¤<Ã¥Ã¾\031Ã¬234Â \237Ãª030Y){lÑ¶"...06Ã'Ãr9Ã¬Ã«Â¸{[Â©\t\205\037\036Ã¿HPH5Âµ\t(Ã©32Ã027Ã¦Ã²036Â¾\"\rÔ¤YÃ215uÂº\027)Ã¶000Ã±215\vÃÃ Ã©Ea\023xÃ©J\206T\204CÃ wr_buf = "\027\003\001\003Â wÂ²'\016Ñ¼Ã033ÃLÃ021Ã)Ã\220hnLÂ¦\020XÃª\236\200\tIhÃÂ¸Sf\234Ã\rÃ ?Â¯Â¼\024FÃ\200Âª/\2264RÃaÃÂ¤IpXÂ»Â¨Ã035Ã¤Qh\211Ã026Ã½4ÃU+\asrÂ£/`Â¤OÃ»1AÊ¹qUÜ¿Â·vÂ±Ã¼\205"... rd = {buf = 0xbf90dfa4 "\026\003\001", pos = 70, used = 70, size = 65536} wr = { buf = 0xbf8fdfa4 "\027\003\001\003Â wÂ²'\016Ñ¼Ã033ÃLÃ021Ã)Ã\220hnLÂ¦\020XÃª\236\200\tIhÃÂ¸Sf\234Ã\rÃ ?Â¯Â¼\024FÃ\200Âª/\2264RÃaÃÂ¤IpXÂ»Â¨Ã035Ã¤Qh\211Ã026Ã½4ÃU+\asrÂ£/`Â¤OÃ»1AÊ¹qUÜ¿Â·vÂ±Ã¼\205"..., pos = 0, used = 0, size = 65536} tls_c = (struct tls_extra_data *) 0xb6208e58 enc_rd_buf = (struct tls_rd_buf *) 0x0 n = 0 flush_flags = <value optimized out> err_src = 0x541a83 "TLS read:" x = <value optimized out> #15 0x08158ade in tcp_read_headers (c=0xb621af88, read_flags=0xbf91e200) at tcp_read.c:406 bytes = -1 remaining = <value optimized out> p = <value optimized out> r = (struct tcp_req *) 0xb621affc #16 0x08158fd4 in tcp_read_req (con=0xb621af88, bytes_read=0xbf91e204, read_flags=0xbf91e200) at tcp_read.c:871 bytes = <value optimized out> total_bytes = 0 resp = <value optimized out> size = <value optimized out> req = (struct tcp_req *) 0xb621affc dst = {send_sock = 0x1, to = {s = {sa_family = 57540, sa_data = "\221Â¿\020\000\000\000\000\000\000\000\020\000\000"}, sin = {sin_family = 57540, sin_port = 49041, sin_addr = {s_addr = 16}, sin_zero = "\000\000\000\000\020\000\000"}, sin6 = {sin6_family = 57540, sin6_port = 49041, sin6_flowinfo = 16, sin6_addr = {in6_u = {u6_addr8 = "\000\000\000\000\020\000\000\000\001\000\000\000\001\000\000", u6_addr16 = {0, 0, 16, 0, 1, 0, 1, 0}, u6_addr32 = {0, 16, 1, 1}}}, sin6_scope_id = 12}}, id = -1080958468, proto = 4 '\004', send_flags = {f = 0 '\0', blst_imask = 0 '\0'}} c = 10 '\n' #17 0x08159a0f in handle_io (fm=0x83296b0, events=1, idx=-1) at tcp_read.c:1150 ret = 4 n = 4 read_flags = 1 con = (struct tcp_connection *) 0xb621af88 s = 12 resp = <value optimized out> t = <value optimized out> #18 0x0815d653 in tcp_receive_loop (unix_sock=35) at io_wait.h:1091 No locals. ---Type <return> to continue, or q <return> to quit--- #19 0x0812885b in tcp_init_children () at tcp_main.c:4819 r = 3 reader_fd_1 = 35 pid = <value optimized out> si = <value optimized out> #20 0x080ad412 in main_loop () at main.c:1632 i = 4 pid = <value optimized out> si = (struct socket_info *) 0x0 si_desc = "udp receiver child=3 sock=192.168.166.31:5060 \000\221Â¿\002\000\000\000/\200\220\000\001\000\000\000\000\000H\000Â¨Ã¤21Â¿\001\000\000\000Â°Â¶Ã¨026Â¶\000\000\000\000\000ÃÂµ\030s2\b\002\000\000\000Ã¢f\b\000ÃÂµ\001\000\000\000\000\000\000\000\001\000\000\000\000\000\000\000Ã¿Ã¿Ã¿Ã¿Â¸Ã¤21Â¿" #21 0x080afbe7 in main (argc=3, argv=dwarf2_read_address: Corrupted DWARF expression. ) at main.c:2398 cfg_stream = (FILE *) 0x8a2d008 c = <value optimized out> r = 0 tmp = 0xbf91e674 "\212Ã¾\221Â¿\233Ã¾\221Â¿Â°Ã¾\221Â¿Â»Ã¾\221Â¿Ã\221Â¿Ã¹1Â¿\020Ã¿\221Â¿DÃ¿\221Â¿LÃ¿\221Â¿WÃ¿\221Â¿]Ã¿\221Â¿oÃ¿\221Â¿{Ã¿\221Â¿\202Ã¿\221 tmp_len = 10492229 port = 136192361 proto = -1080957480 ret = <value optimized out> seed = 21782655 rfd = 4 debug_save = <value optimized out> debug_flag = 0 dont_fork_cnt = 0 n_lst = <value optimized out> p = <value optimized out> Best Regards

4 24

[Kamailio-Users] BYE message not relayed to caller
by Vikram Ragukumar 04 Dec '12

04 Dec '12

Hello, My setup includes a cellphone that uses Wi-Fi to hookup to the internet with IP 192.168.1.101. Cellphone app, tries to register with VOIPSwitch through an outbound proxy server(Kamailio + rtpproxy), which rewrites SDP to force rtp through an rtpproxy. I can make a call from the cellphone to a PSTN number. When the PSTN endpoint hangs up, VOIPSwitch relays a BYE message to the proxy that looks like : No. Time Source Destination Protocol Info 27 18.383929 IPaddr_VoipSwitch IPaddr_proxy SIP Request: BYE sip:1234@IPaddr_VoipSwitch Frame 27 (420 bytes on wire, 420 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: IPaddr_VoipSwitch , Dst: IPaddr_proxy User Datagram Protocol, Src Port: sip (5060), Dst Port: 7160 (7160) Session Initiation Protocol Request-Line: BYE sip:1234@IPaddr_VoipSwitch SIP/2.0 Method: BYE [Resent Packet: False] Message Header Route: <sip:IPaddr_proxy:7160;lr=on;nat=yes> CSeq: 1 BYE Via: SIP/2.0/UDP IPaddr_VoipSwitch:5060; branch=z9hG4bk180242100434182912333954 From: sip:5678@IPaddr_VoipSwitch;tag=18024210041929123187506289 Call-ID: LUOyIDm-ecy36NcUfj5G3v1lM6g5snkM To: "1234" <sip:1234@IPaddr_VoipSwitch>;tag=9XHJmiuf58vTrF7.vwflEK-G63JVPQSi Content-Length: 0 The request URI in the above message is the IP address of VOIPSwitch itself, which causes the proxy to forward the BYE message back to VOIPSwitch. The cellphone never receives the BYE message and consequently never hangs up. Any suggestions on how i can get around this problem ? Thanks and Regards, Vikram.

3 9

is kamilio support RFC5626 now ?
by mike 16 Oct '12

16 Oct '12

dear kamilio developers : i read the kamilio 3.3.0' s new features , it say it support outbound , and i have place to handle reg-id and sip.instance . so , i wonder if the kamilio3.3.0 support RFC5626 now，so that the kamilio can be used as a edge proxy . i ask why because i can't find where kamilio add flow-token and path for the register request. thanks

5 8

kamailio core at qm_status
by Jijo 01 Oct '12

01 Oct '12

Hi All, I'm observing a core intermittently at "qm_status (qm=0x786cd000) at mem/q_malloc.c:763" for kamailio version 3.1.0 looking at the backtrace this is occurring while doing a tcp buffer overrun. Please have a look and let me know if anyone observed this issue or how can we debug it. Please find the backtrace and kamailio version osbprod-V2R0:~ # /usr/sbin/kamailio -V version: kamailio 3.1.0 (i386/linux) 21a375 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 15MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 21a375 compiled on 12:38:44 Apr 26 2012 with gcc 4.5.0 #0 qm_status (qm=0x786cd000) at mem/q_malloc.c:763 #1 0x0819c0f3 in qm_debug_frag (qm=0x786cd000, f=<value optimized out>) at mem/q_malloc.c:167 #2 0x0819c352 in qm_free (qm=0x786cd000, p=0x7a6cb6ac, file=0x821cce2 "<core>: tcp_main.c", func=0x822285f "wbufq_run", line=880) at mem/q_malloc.c:457 #3 0x0815f686 in wbufq_run (tcpconn=0x7a1a7548, ev=4, fd_i=-1) at tcp_main.c:880 #4 handle_tcpconn_ev (tcpconn=0x7a1a7548, ev=4, fd_i=-1) at tcp_main.c:4141 #5 0x08169bf2 in io_wait_loop_epoll () at io_wait.h:1092 #6 tcp_main_loop () at tcp_main.c:4606 #7 0x080b0404 in main_loop () at main.c:1655 #8 0x080b1f84 in main (argc=9, argv=0xbfb4c9f4) at main.c:2446 (gdb) l 758 LOG_(DEFAULT_FACILITY, memlog, "qm_status: ", 759 "dumping free list stats :\n"); 760 for(h=0,i=0;h<QM_HASH_SIZE;h++){ 761 unused=0; 762 for (f=qm->free_hash[h].head.u.nxt_free,j=0; 763 f!=&(qm->free_hash[h].head); f=f->u.nxt_free, i++, j++){ 764 if (!FRAG_WAS_USED(f)){ 765 unused++; 766 #ifdef DBG_QM_MALLOC 767 LOG_(DEFAULT_FACILITY, memlog, "qm_status: ",

4 7

[Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers
by Samuel Muller 26 Sep '12

26 Sep '12

Hello all, I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose is to have several interconnections with PSTN. I configured it like this : Audiocodes registers as a gateway to the Kamailio, using a dedicated port (5062). Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the proxy. I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm. But the audiocodes returns some errors about SIP headers sent by Kamailio : ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected Here you have the example of an INVITE from a SIP phone to the PSTN : ** audiocodes debug ** 4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from 77.246.81.132:5060 ---- INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0 Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes> Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0 Via: SIP/2.0/UDP 192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96 From: "Sam" <sip:0123451010@sip.720.fr <sip%3A0123451010(a)sip.720.fr>>;tag=71078b346a20fb3eo0 To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr>> Call-ID: 944d8aec-27503ee6(a)192.168.0.113 CSeq: 102 INVITE Max-Forwards: 49 Contact: "Sam" <sip:0123451010@77.246.81.162:15170> Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 281 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER Supported: 100rel, replaces Content-Type: application/sdp P-Asserted-Identity: <0123451010> Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes v=0 o=- 26933860 26933860 IN IP4 192.168.0.113 s=- c=IN IP4 77.246.81.133 t=0 0 m=audio 35038 RTP/AVP 18 0 8 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=nortpproxy:yes ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected ( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26] ( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26] ( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26] The outgoing INVITE from Kamailio is exactly the same received by the AudioCodes. When I searched over Google, I just found 2 answers about Asterisk / Audiocodes unsolved problem, but no more informations. I supposed that the problem is as indicated : " s=- " where source is empty in place of "NULL" / "0" or something like this ... Someone can confirm or already met the problem ? Many thanks all :) .Sam.

5 5

[Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks
by Joe Hart 15 Sep '12

15 Sep '12

Hi all, For a project on which I'm currently working, I am having some problems figuring out how to correctly configure Kamailio to communicate with RTP Proxy in order to send media into and out of a network with private IP address ranges. I have a proxy set up to send the SIP traffic, and all of this is working fine. However, I'm having some trouble getting the RTP Proxy set up. Currently, when the call is connected, the offer/answer is made and RTP Proxy seems to be taking over, but I'm having trouble getting my audio to flow in both directions. Examination of the traffic coming into and out of this machine seems to indicate that the IP addresses aren't being mangled correctly. Specifically, it appears the internal IP address isn't being changed to reflect the IP address of the machine on which RTP Proxy is running, so that when the caller tries to send audio back, the IP it's given to reply to is 10.10.x.x, which obviously won't work. I have tried experimenting with specifically setting IP addresses in the rtpproxy_offer() and _answer() methods to no avail, as well as setting various flags in those methods. However, I must admit that I'm not entirely sure what's happening under the hood with these methods, or what rtpproxy is doing with that information when it gets it. Rather than continue to hack at this by trial and error, I'm hoping someone here can point me in the right direction. Any advice, example code or pep talks would be greatly appreciated. Thanks in advance, -- Joe Hart Voice Systems Integrator Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0673

8 37

Suggestions for time utils
by Andreas Granig 16 Aug '12

16 Aug '12

Hi, I'd like to implement a couple of helper functions for time handling, e.g. checking day of week, day of month etc. from within the kamailio config file. What would you prefer, a new module ("timeutils" maybe?) or adding it to cfgutils (there are already time-based functions there like sleep and usleep)? The idea is to implement time based call-forwards, and an approach could be to provision various time-related values in usr_preferences table, and then check it in the config. For example, when doing call-forwards from Monday to Friday only, I could put this into usr_preferences: attribute: cf_weekday value: [1, 2, 3, 4, 5] (each entry is a separate row in usr_preferences) And in kamailio config, I'd call this: avp_db_load(...); if(is_weekday("$avp(s:cf_weekday)")) { do CF } So is_weekday would iterate over the entries in the avp list and return true if the weekday at the time of routing matches an entry in the list. A module config param could control whether to use gmtime or localtime for matching. Does this make sense? Suggestions for other approaches? I'd rather prefer to do it directly in config instead of using some external interpreter like lua, python etc. Andreas

3 17

Possible bad lcr module, after upgrade to 3.3.0 from 3.2
by Alexey Mechanoshin 13 Aug '12

13 Aug '12

Hello. After upgrade kamailio and db, lcr module bad routing if entered *any value*to from_uri column in lcr_rule table: mysql> select * from lcr_rule where id=5; +----+--------+--------+-------------------------------+-------------+---------+---------+ | id | lcr_id | prefix | from_uri | request_uri | stopper | enabled | +----+--------+--------+-------------------------------+-------------+---------+---------+ | 5 | 1 | yyyy | sip:x[xx]\d{2}@xxx.xxx.xx.xx | NULL | 0 | 1 | +----+--------+--------+-------------------------------+-------------+---------+---------+ and load_gw(...) skip this rule, in log: INFO: lcr [lcr_mod.c:1493]: skipping disabled <gw/rule> = <1/5> If remove value from 'from_ui' - rule enabled and all work. Why is turned off rule ? Linux Debian squeeze, upgraded from "deb http://deb.kamailio.org/kamailiosqueeze main" -- "В связи с отсутствием интереса общественности, конец света отменяется" (с) ? С Уважением, Механошин Алексей

3 11

Outbound and Registrar. No binding update
by José Luis Millán 02 Aug '12

02 Aug '12

Hi, I'm using Kamilio 3.3.0 as registrar server. I`m using an outbound proxy so 'use_path' parameter or 'registrar' module is enabled. According to RFC 5626, a re-registration from a specific combination of AoR, instance_id and reg_id must update the binding. "" If the registrar receives a re-registration for a specific combination of AOR, and instance-id and reg-id values, the registrar MUST update any information that uniquely identifies the network flow over which the request arrived if that information has changed, and SHOULD update the time the binding was last updated. "" In my installation this is not fullfilled as shown: "" AOR:: jmillan Contact:: sip:jmillan@MY_IP;transport=ws;ov-ob=a570655c14 Q= Expires:: 181 Callid:: 1hgq3khalq2rzfr Cseq:: 108 User-agent:: JsSIP 0.1.0 Path:: <sip:OUTBOUND_IP:9090;transport=tcp;lr;ovid=de0c0b9f>,<sip:a570655c14@OUTBOUND_IP :10080;transport=ws;lr;ovid=de0c0b9f;ob> State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:KAM_IP:5060 Methods:: 783 Ruid:: uloc-50119d99-328e-1 Instance:: <urn:uuid:38dce009-ae1f-4fd1-91dc-99ed9affddc1> Reg-Id:: 1 Contact:: sip:jmillan@MY_IP;transport=ws;ov-ob=1dd97b4d51 Q= Expires:: 193 Callid:: 1hgq3khalq2rzfr Cseq:: 110 User-agent:: JsSIP 0.1.0 Path:: <sip:OUTBOUND_IP:9090;transport=tcp;lr;ovid=de0c0b9f>,<sip:1dd97b4d51@OUTBOUND_IP :10080;transport=ws;lr;ovid=de0c0b9f;ob> State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:KAM_IP:5060 Methods:: 783 Ruid:: uloc-50119d99-3290-2 Instance:: <urn:uuid:38dce009-ae1f-4fd1-91dc-99ed9affddc1> Reg-Id:: 1 "" The scenario is such that when the client looses the connection with the Outbound Server, it reconnects and re-registers to Kamailio in order to replace registration and be able to receive in-dialog messages. Any comment is wellcome. Regards,

4 19

planning release of v3.3.1
by Daniel-Constantin Mierla 02 Aug '12

02 Aug '12

Hello, I am planning to package v3.3.1 out of the latest branch 3.3 this Thursday, August 2. If you are aware of issues that are not listed on the tracker (http://sip-router.org/tracker/), add them there to see what can be sorted out before. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 1

cannot allocate memory & increase shared memory
by Alex Solt 01 Aug '12

01 Aug '12

Hi, We are getting "cannot allocate memory" error in the log file. I was wondering which one is the right way to increase the shared memory: 1) adding the following to /etc/openser/openserctl STARTOPTIONS="-m 128" 2) modify the following in /etc/default/openser MEMORY=1024 Thanks,AS

2 1

New Wiki Site has problems?
by Andrew Pogrebennyk 31 Jul '12

31 Jul '12

At least some links at http://www.kamailio.org/wiki/ don't work at the moment: Install Kamailio v3.3.x From GIT Upgrade Kamailio v3.1.x to v3.2.0 Upgrade Kamailio v3.2.x to v3.3.0 you are redirected to some instruction from DokuWiki Installer when trying to visit them. Could somebody please check it? Thanks. Andrew

2 1

routing based on username
by Richard Zheng 31 Jul '12

31 Jul '12

Hi, We are trying to follow http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdbto setup Kamailio as front end and multiple PBXs as back end. We need to forward sip messages to the respective servers based on the contact. For example, users 101, 102 go to PBX at 1.1.1.1 and users 201, 202 go to PBX at 1.1.1.2. Where is a good place to save this info and how to route them in the config? Also we have some users with fixed IP addresses which don't use authentication. How do we route the calls based on source IP? Since we have plenty of these users, we'd like to save them in database too. Any advice is appreciated! Richard

3 2

Add Own Plug-in
by Ananth Kollipara 31 Jul '12

31 Jul '12

Hi, Can you please provide the steps to add a plug-in? Regards, Ananth

2 1

ACK not relayed
by David Notivol 31 Jul '12

31 Jul '12

In a UAC-Kamailio-UAS scenario, we've found a case where the ACK coming from uac is not relayed by our proxy to the uas. This is the log for the ACK message: Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:624]: SIP Request: Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:626]: method: <ACK> Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:628]: uri: <sip:59138553189@79.170.68.157> <sip:59138553189@79.170.68.157> Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:630]: version: <SIP/2.0> Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 235, <rport> = <n/a>; state=6 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bKKc4gKXyemS9HD>; state=16 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=2 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: this is the first via Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [receive.c:149]: After parse_msg... Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [receive.c:190]: preparing to run routing scripts... Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: sl [sl_funcs.c:396]: DEBUG : sl_filter_ACK: to late to be a local ACK! Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: maxfwd [mf_funcs.c:85]: value = 65 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=6ne1x6d6-CC-23 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:188]: DEBUG: get_hdr_field: <To> [52]; uri=[sip:59138553189@79.170.68.157] Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: to body [ <sip:59138553189@79.170.68.157> <sip:59138553189@79.170.68.157>] Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:168]: get_hdr_field: cseq <CSeq>: <31362871> <ACK> Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:202]: DEBUG: get_hdr_body : content_length=0 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:104]: found end of header Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=0cS8trtaF196F Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: siputils [checks.c:106]: totag found Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: ERROR: <script>: ESTAMOS EN WITHIN Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: rr [loose.c:108]: No Route headers found Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: rr [loose.c:829]: There is no Route HF Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: ERROR: <script>: NO LOOSE ROUTE However, in a pcap trace I clearly see the Route header: This is the Record-Route in the 200 message Record-Route: <sip:79.170.68.157;lr;ftag=330cdf48-co1022-INS033> And this is the Route header in the ACk: Route: <sip:79.170.68.157;lr;ftag=330cdf48-co1022-INS033> I haven't been able to figure out what's going on. We have plenty of traffic working in that proxy. What am I missing? Thanks in advance David.

2 8

BYE Race Condition
by Brandon Armstead 31 Jul '12

31 Jul '12

Hello, I am running into an issue where there is a race condition happening. I am looking for opinions / ideas on how to handle the following below scenario. Scenario. UAC places an outbound call -> upstream carrier. The call is disconnected on both ends at the exact same time, UAC -> sends BYE upstream Upstream Carrier -> sends BYE downstream Upstream 200 OK's the BYE UAC sends 481 back to Upstream Carrier for their generated BYE. The upstream carrier is complaining about receiving the relayed 481 responses -- so my first thought was simply to drop() these from relaying upstream. I am curious how other people are handling this? Would you suggest simply dropping the relay from being sent back upstream on the 481? Would you simply always 200 OK a downstream BYE from trusted carriers regardless of UAC response, and create separate transaction to send BYE downstream? Thank you as always. Look forward to your thoughts / suggestions / ideas. Sincerely, Brandon Armstead

2 2

Error 403 not relaying
by Tayeb Meftah 30 Jul '12

30 Jul '12

hello, i'm using Kamailio with a simple registrare configuration i used the apt repository to install the developmant package of kamailio domain sip.tmvoip.net have 2 records, A record and AAA record (dual stacked) i configured it in my kamailio using siremis and in the kamctlrc config below: Kamailio.cfg: http://paste.debian.net/180972/ Kamctlrc: http://paste.debian.net/180973/ siptrace: http://paste.debian.net/180974/ note that 172.28.0.0/21 and 41.221.20.12/30 is routed, no nat. my current issue is if i register or i invite, i get error 403 not relaying trying to investigate this issu but i have no clue thank you everyone, -- Voice of the blind ™ broadcast freedom Phones: Algeria: +213-660-347746 USA, NewYork: +1(347)759-5883 iNum: +883510001288000 we're blind, but not deaf!

3 3

Re: [SR-Users] Can't initialize shared memory
by Uri Shacked 30 Jul '12

30 Jul '12

> > this is the log when debug is 4... > > Jul 24 10:19:37 RNDSRV kamailio: : <core> [cfg.y:3532]: parse error in > config file /usr/local/etc/kamailio/global_params.cfg, line 121, column 54: > Can't initialize shared memory > Jul 24 10:19:37 RNDSRV kamailio: ERROR: mi_fifo [mi_fifo.c:271]: FIFO stat > failed: Bad address > > > not much. > line 121 on the file is the modparam for mi_fifo indication the > /tmp/kamailio_fifo file. > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > any ideas? > > >

2 1

Reply to one topic correctly
by Duong Manh Truong 30 Jul '12

30 Jul '12

Hi list, I am a newbie of the maillist, I'm wondering that whenever i reply to a specific topic with the subject of "RE: [SR-Users]....", my email come to maillist in the way dislike what i wish, my email appears like a NEW topic instead of SUB topic (reply message form) I try several ways but still not get the correct method of reply, please help me to fix this, thanks.

2 1

Re: [SR-Users] Kamailio with APNS
by Duong Manh Truong 29 Jul '12

29 Jul '12

Thank Klaus, I've try with t_suspend() and t_continue() as your guide, *#Failure route to react with negative messages failure_route[1] { if (t_check_status("5[0-9][2-9]")) { if(t_suspend()) { xlog("proceesing suspended in trasaction [$T(id_index):$T(id_label)]\n"); xlog("before sleep \n"); } if(async_sleep("20")) xlog("After sleep 20s \n"); if(t_continue('1','1','MYROUTE')) xlog("Continuation successfully \n"); exit; };* And i found that t_suspend() and async_sleep() runnung ok but t_continue() seems not running. I read about TMX module and TMX module Pseudo-Variables but still can not figure out it :( Secondly, in case of t_continue() runs successfully , please tell me how to let Kamailio resend the INVITE message to client B (before the t_continue(), i think so ) - while client B is waking up ? Which functions and modules that i need to do this? Thanks. ----------------------------------------------------------------- Hi! You have to use a failure route (http://kamailio.org/docs/modules/3.3.x/modules/tm.html#t_on_failure). In the failure route check if the reply code is 404 and if yes, trigger the APNS logic and then put the call e.g. on a stack and resume e.g. every second and ofrward to the "SIP server" again to see if the client is registered meanwhile. For the asynchronous processing take a look at these functions: t_suspend(),t_resume()[1]. For example you can call t_resume in a timer route every second. For the timer route, to know which transactions are currently suspended and are waiting for further processing, you could put the transaction identifiers in a message-queue [2] and read the queue from the timer route. regards Klaus [1] http://kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2535968 [2] http://kamailio.org/docs/modules/3.3.x/modules/mqueue.html On 24.07.2012 06:02, Duong Manh Truong wrote: >* Hi list,*>**>* My team are developing a SIP client for mobile running iOS ,*>**>* then having to deal with the "back ground mode" problem while this soft*>* is installed on mobiles*>**>* One of the solutions is using APNS. (Apple Push Notification Service)*>**>* I wanna to use Kamailio as SIP Redirect / Proxy for my current SIP server*>**>* The model is like that:*>**>* User A <---------> Kamailio <---------> SIP server <--------> User B*>**>* IF UserA call UserB, UserB is not available now (not registered yet),*>* normally, SIP server will reply Kamailio with "404 Not found"*>**>* In that case, i want to set up Kamailio do the communication with APNS*>* and waiting for UserB "Wake up" then RESEND the Invite message ......*>* (instead of forwarding this 404 message to UserA)*>**>* I am getting stuck at how Kamailio can control "404 Not found" message*>* which it receives from SIP server .*>**>* I do searching around several days but still can not get any clearer*>* points.*>**>* Please help me with some instruction/ configuration of Kamailio for this*>* case !*>**>* Thank you very much.*>**>**>* _______________________________________________*>* SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list*>* sr-users at lists.sip-router.org <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>* http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users*>**

1 0

How to change _only_ next hop and dpn't touch content of message?
by intel＠intrans.baku.az 29 Jul '12

29 Jul '12

I want retransmit some me ssages to other sip server as is. f.e server sip1.com receives invite message INVITE user(a)somewhere.org i want to send it to server sip2.com as is. Unfortunately, if i use rewritehostport() function, or modify $rp $rd pseudovariables fe $rp='sip2.com', kamailio changes invite uri, and send to sip2.com INVITE user(a)sip2.com The To: filed left untouched How can I resend message without modifications? I've tried also modify $dp and $dd pseudovariables, but then message wasn't sent anywhere,and in log I can see this message: Jul 27 15:44:54 dsdb01 /usr/sbin/kamailio[17786]: ERROR: tm [t_fwd.c:1523]: ERROR: t_forward_nonack: no branches for forwarding Jul 27 15:44:54 dsdb01 /usr/sbin/kamailio[17786]: ERROR: tm [tm.c:1375]: ERROR: w_t_relay_to: t_relay_to failed What else can I try? --

2 1

BYE Race Condition
by Brandon Armstead 28 Jul '12

28 Jul '12

*** Please forgive if this is a duplicate *** Hello, I am running into an issue where there is a race condition happening. I am looking for opinions / ideas on how to handle the following below scenario. Scenario. UAC places an outbound call -> upstream carrier. The call is disconnected on both ends at the exact same time, UAC -> sends BYE upstream Upstream Carrier -> sends BYE downstream Upstream 200 OK's the BYE UAC sends 481 back to Upstream Carrier for their generated BYE. The upstream carrier is complaining about receiving the relayed 481 responses -- so my first thought was simply to drop() these from relaying upstream. I am curious how other people are handling this? Would you suggest simply dropping the relay from being sent back upstream on the 481? Would you simply always 200 OK a downstream BYE from trusted carriers regardless of UAC response, and create separate transaction to send BYE downstream? Thank you as always. Look forward to your thoughts / suggestions / ideas. Sincerely, Brandon Armstead

1 0

lcr.dump_gws problem
by Gary Chen 27 Jul '12

27 Jul '12

Kamailio 3.2.0 When testing Kamailio 3.2.0, I notice that lcr.dump_gws command does not display ip_addr correctly. Here is my output: lcr_id: 3 gw_id: 15 gw_index: 2 gw_name: gateway_1 scheme: sip ip_addr: 1273060816.0.0.0 hostname: port: 5060 params: strip: 0 prefix: tag: flags: 0 defunct_until: 0 LCR is still working correctly. It just does not display IP in the right form. Does any body also has the same problem? Gary

3 4

Re: [SR-Users] Add CallerID to SIP header
by Dominic Raferd 27 Jul '12

27 Jul '12

1 0

Add CallerID to SIP header
by Dominic Raferd 27 Jul '12

27 Jul '12

I'm not yet using Kamailio. I am looking for something to do a job which sounds really simple but may not be. Our PBX connects to a SIP service (voicetrading.com) but does not supply our CallerID, this means that voicetrading.com provide some very weird callerids for the person at the other end (international numbers etc). Which is not good. There is no way to get our PBX to add CallerID itself, so I thought maybe I could set up a local sip proxy? or sip redirect? server (Kamailio?) to do nothing other than add in the callerid details to the sip header and then forward the sip message on to voicetrading i.e. be otherwise 'invisible'? If Kamailio is not the right thing to do this with, is there something else that could do it? Thanks for any suggestions. Dominic

2 1

Cant make from_uri, request_uri work for LCR module
by Antanas Masevicius 27 Jul '12

27 Jul '12

Hello, having trouble running LCR in kamailio 3.3 (head). Here is my DB excerpt which i use to reproduce the problem: INSERT INTO `lcr_rule` VALUES (2617960,90,'118','.*',1,1,'.*'); INSERT INTO `lcr_rule_target` VALUES (7771454,90,2617960,1424,0,100); INSERT INTO `lcr_gw` VALUES (1424,90,'OP1','10.10.10.10','',5060,NULL,1,0,NULL,'11#',1,NULL,NULL); I stripped all other LCR data from my DB and only this rule with from_uri and request_uri seems to bother me. Tried this version on debian lenny and debian squeeze. On Lenny i am not able to load lcr tables at all, get a segfault. On squeeze, lcr tables loads up, but this rule gets marked as disabled. Here is config part which i use to load a rule: if (load_gws(90, $rU, $fu)) { if (next_gw()) { t_on_failure("FAIL_GW"); } else { route(NO_ROUTE); } On squeeze, if i add following code to lcr_mod.c:1249 : LM_INFO("LOADING lcr rule <%u> \n", rule_id); right after: rule_id = (unsigned int)VAL_INT(ROW_VALUES(row)); rule_id is still printed, but i get segfault like this: Jul 24 10:17:43 sp1 kernel: [23482626.105865] kamailio[15136]: segfault at a4 ip 00007fde15cb8802 sp 00007fff39d68320 error 4 in lcr.so[7fde15ca8000+23000] Not sure what i do terribly wrong. Setting from_uri and request_uri to anythig but NULL makes the code to segfault with that print line added. Without print line, kamailio starts, but never loads that rule. Any help would be appreciated. best regards, Antanas Masevicius NTT

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Re: [SR-Users] Cant make from_uri, request_uri work for LCR module
by Alex Balashov 26 Jul '12

26 Jul '12

When calling load_gws(), try putting $rU and $fu in double quotes, e.g. "$rU" rather than $rU bare. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Antanas Masevicius <antanas.masevicius(a)ntt.lt> wrote:Hello, having trouble running LCR in kamailio 3.3 (head). Here is my DB excerpt which i use to reproduce the problem: INSERT INTO `lcr_rule` VALUES (2617960,90,'118','.*',1,1,'.*'); INSERT INTO `lcr_rule_target` VALUES (7771454,90,2617960,1424,0,100); INSERT INTO `lcr_gw` VALUES (1424,90,'OP1','10.10.10.10','',5060,NULL,1,0,NULL,'11#',1,NULL,NULL); I stripped all other LCR data from my DB and only this rule with from_uri and request_uri seems to bother me. Tried this version on debian lenny and debian squeeze. On Lenny i am not able to load lcr tables at all, get a segfault. On squeeze, lcr tables loads up, but this rule gets marked as disabled. Here is config part which i use to load a rule: if (load_gws(90, $rU, $fu)) { if (next_gw()) { t_on_failure("FAIL_GW"); } else { route(NO_ROUTE); } On squeeze, if i add following code to lcr_mod.c:1249 : LM_INFO("LOADING lcr rule <%u> \n", rule_id); right after: rule_id = (unsigned int)VAL_INT(ROW_VALUES(row)); rule_id is still printed, but i get segfault like this: Jul 24 10:17:43 sp1 kernel: [23482626.105865] kamailio[15136]: segfault at a4 ip 00007fde15cb8802 sp 00007fff39d68320 error 4 in lcr.so[7fde15ca8000+23000] Not sure what i do terribly wrong. Setting from_uri and request_uri to anythig but NULL makes the code to segfault with that print line added. Without print line, kamailio starts, but never loads that rule. Any help would be appreciated. best regards, Antanas Masevicius NTT _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

5 8

siremis errors
by Tony Ferrario 26 Jul '12

26 Jul '12

I just installed a Siremis 3.2.1 on a server with an already working Kamailio 3.2.1. bellow is a typical error message I keep getting. Can you point me to a solution -Thanks [http://192.168.105.196/siremis/themes/default/images/icon_forbidden.gif] System Internal Error The detailed error message are below: [2012-07-26 19:14:11 (GMT)] An exception occurred while executing this script: Error message: #0, Error in query:: SELECT T0.id, T0.method, T0.from_tag, T0.to_tag, T0.callid, T0.sip_code, T0.sip_reason, T0.time, T0.src_ip, T0.dst_user, T0.dst_domain, T0.dst_ouser, T0.src_user, T0.src_domain, T0.cdr_id FROM `acc` T0 WHERE method='INVITE' AND T0.src_user='admin' LIMIT 10. SQLSTATE[42S22]: Column not found: 1054 Unknown column 'T0.src_ip' in 'field list' Script name and line number of error: /var/www/siremis-3.2.1/openbiz/bin/data/BizDataObj_Lite.php:447 function: _run_search ( Array(2) ) @ /var/www/siremis-3.2.1/openbiz/bin/data/BizDataObj_Lite.php 243 function: fetch ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyForm.php 690 function: fetchDataSet ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/FormRenderer.php 69 function: renderSmarty ( Object(EasyForm), "/var/www/siremis-3.2.1/siremis/modules/sipuser/template/grid.tpl" ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/FormRenderer.php 45 function: render ( Object(EasyForm) ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyForm.php 1648 function: renderHTML ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyForm.php 1538 function: render ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/ViewRenderer.php 83 function: renderSmarty ( Object(EasyView), "/var/www/siremis-3.2.1/siremis/modules/sipuser/template/view.tpl" ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/ViewRenderer.php 40 function: render ( Object(EasyView) ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyView.php 274 function: _render ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyView.php 237 function: render ( ) @ /var/www/siremis-3.2.1/openbiz/bin/BizController.php 221 function: renderView ( "sipuser.view.AccListView", "", "", Null, "" ) @ /var/www/siremis-3.2.1/openbiz/bin/BizController.php 107 function: dispatchRequest ( ) @ /var/www/siremis-3.2.1/openbiz/bin/BizController.php 32 function: include_once ( "/var/www/siremis-3.2.1/openbiz/bin/BizController.php" ) @ /var/www/siremis-3.2.1/siremis/bin/controller.php 6 function: include ( "/var/www/siremis-3.2.1/siremis/bin/controller.php" ) @ /var/www/siremis-3.2.1/siremis/bin/_forward.php 102 function: include ( "/var/www/siremis-3.2.1/siremis/bin/_forward.php" ) @ /var/www/siremis-3.2.1/siremis/index.php 3 ________________________________ Please ask system administrator for help... Tony Ferrario Manager SVT & Ops OnMobile Global Limited 3950 Civic Center Drive , Suite 200 San Rafael, Ca 94903 Phone: 415-223-8206 Cell: 707-508-7285 tony.ferrario(a)onmobile.com ________________________________ DISCLAIMER: The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. Further, this e-mail may contain viruses and all reasonable precaution to minimize the risk arising there from is taken by OnMobile. OnMobile is not liable for any damage sustained by you as a result of any virus in this e-mail. All applicable virus checks should be carried out by you before opening this e-mail or any attachment thereto. Thank you - OnMobile Global Limited.

1 0

Overload Control
by Kaplan, Michael A 26 Jul '12

26 Jul '12

Folks, I'm trying to figure out if either Kamailio or OpenSIPs provides the various overload control algorithms recommended in the IETF or elsewhere. I haven't found documentation to this effect. Has anyone looked into this or used such techniques? Thanks, Mike

3 6

rtimer execution period - round two
by Alex Balashov 25 Jul '12

25 Jul '12

Hi Daniel, I wanted to go back to this thread for a minute: http://lists.sip-router.org/pipermail/sr-users/2012-June/073663.html Sorry that I didn't reply to the thread; I no longer have the last message. In your reply, you said: > there is no timeout on the execution of that task. Of course, you > can start many rtimer processes to do same task, up to you, > via configuration parameters. A matter of how rtimer executed > route is built, a rtimer process may not rest (e.g., sleep) and > keep consuming mqueue items as long as the queue is not empty." However, I am seeing some behaviour that is rather different. Roughly, I have a single rtimer task that runs every 3 seconds and consumes an mqueue full of accounting events. The rtimer request route looks like something this: route[ACCT_WRITER] { while(mq_fetch("acct_write")) { # Some stuff to write $mqv(...) values to PostgreSQL. ... sql_query("ca", "SELECT * FROM acct_event_proc(...)"); ... mq_pv_free("account_write"); } } And the rtimer task is parameterised thusly: modparam("rtimer", "timer", "name=acct_write;interval=3;mode=1") modparam("rtimer", "exec", "timer=acct_write;route=ACCT_WRITER") And the mqueue: modparam("mqueue", "mqueue", "name=acct_write") I have a stored procedure - let's call it acct_event_proc() - that is invoked from within the rtimer task. When I looked at the live query activity from Kamailio to Postgres by doing a packet capture on loopback, like this: tcpdump -i any -A -s 0 -tttt -n 'tcp port 5432' | grep acct_event_proc What I saw was some events being written for about ~3 seconds and then the process would stop. Then, after about half a second, it would burst again for 3 seconds, then stop again. And so on and so forth. When I lowered the interval to 1 second, it got faster and more responsive, and would push more events through in one interval. But still, it appeared to be a limited "burst" of SELECT calls punctuated by 1 second of silence. When I checked out branch 3.3 and lowered the interval to 50000u, it got even faster, almost real-time enough to keep up. Still, I did not really know what was happening, so I added an MI command to HEAD to check mqueue size live: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commit;h=41f771… (I think it would be quite useful anyhow, as there is no runtime mqueue info available right now.) Using it, I confirmed my impression: the bigger the rtimer interval, the larger the queue grows, and the slower the events seem to be written overall. The shorter the rtimer interval, the faster they are dequeued, and the smaller the mqueue is overall, in the course of normal call processing. Thus, my impression is that the execution time of the rtimer task is bounded by the interval, and that the core kills the forked rtimer task currently running prior to starting another one. Can you shed some light? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

3 12

Cleanup location table on failed NAT ping
by David 25 Jul '12

25 Jul '12

Hello, I am using kamailio 3.2.3. I am using the nathelper module to "ping" users in the location table to keep the NAT bindings alive. In the event of 3 failed NAT pings, I would like to mark the entry as "unavailable" or completely delete it from the location table to avoid having stale data. Is this a feature that exists ? Is not, are their plans for it in the future ? When the REGISTER expires after 3600, if the phone doesn't register, the location entry gets cleaned up. So I have to wait up to one hour before the bad entry is purged. I was hoping to have it purged sooner. Thanks, David version: kamailio 3.2.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 14:19:23 Apr 19 2012 with gcc 4.4.3

3 2

Error Loading db_postgres.so on Startup
by Akan 25 Jul '12

25 Jul '12

I am getting this error when trying to run kamailio on Centos 5.5. I have Postgresql 9.1.4 installed and I have tried with both OpenSSL v0.9.8e and v1.0.1c and I still get the same error. Any help would be appreciated. ERROR: <core> [sr_module.c:565]: ERROR: load_module: could not open module </usr/local/kamailio-3.2/lib64/kamailio/modules/db_postgres.so>: /usr/local/kamailio-3.2/lib64/kamailio/modules/db_postgres.so: undefined symbol: TLSv1_method Thanks Nathaniel

2 4

Kamailio with APNS
by Duong Manh Truong 24 Jul '12

24 Jul '12

Hi list, My team are developing a SIP client for mobile running iOS , then having to deal with the "back ground mode" problem while this soft is installed on mobiles One of the solutions is using APNS. (Apple Push Notification Service) I wanna to use Kamailio as SIP Redirect / Proxy for my current SIP server The model is like that: User A <---------> Kamailio <---------> SIP server <--------> User B IF UserA call UserB, UserB is not available now (not registered yet), normally, SIP server will reply Kamailio with "404 Not found" In that case, i want to set up Kamailio do the communication with APNS and waiting for UserB "Wake up" then RESEND the Invite message ...... (instead of forwarding this 404 message to UserA) I am getting stuck at how Kamailio can control "404 Not found" message which it receives from SIP server . I do searching around several days but still can not get any clearer points. Please help me with some instruction/ configuration of Kamailio for this case ! Thank you very much.

3 3

Kamailio with APNS
by Duong Manh Truong 24 Jul '12

24 Jul '12

Hi list, My team are developing a SIP client for mobile running iOS , then having to deal with the "back ground mode" problem while this soft is installed on mobiles One of the solutions is using APNS. (Apple Push Notification Service) I wanna to use Kamailio as SIP Redirect / Proxy for my current SIP server The model is like that: User A <---------> Kamailio <---------> SIP server <--------> User B IF UserA call UserB, UserB is not available now (not registered yet), normally, SIP server will reply Kamailio with "404 Not found" In that case, i want to set up Kamailio do the communication with APNS and waiting for UserB "Wake up" then RESEND the Invite message ...... (instead of forwarding this 404 message to UserA) I am getting stuck at how Kamailio can control "404 Not found" message which it receives from SIP server . I do searching around several days but still can not get any clearer points. Please help me with some instruction/ configuration of Kamailio for this case ! Thank you very much.

2 1

Re: [SR-Users] Kamailio with APNS
by Nguyen Thanh Tien 24 Jul '12

24 Jul '12

I think you can find the 404 message at the onreply_route[]. I tried and found the 200 OK here.

1 0

SIREMIS
by Adam Fyfe 24 Jul '12

24 Jul '12

Hi, I am testing Siremis, and have become stuck. How does one add an outbound SIP trunk for calls ? thanks

2 1

Kamailio setup problem
by Vijay Thakur 23 Jul '12

23 Jul '12

All Experts, I have installed Kamailio Server on Linode Cloud. I have followed the following URL for Setup: http://kb.asipto.com/kamailio:presence:k31-made-simple This is done in ubuntu 10.04. I am only able to make chat. Video chat, conference, Presence, Call, Screen Sharing is not working. But followed the same how to install kamailio in my local system and all features are working fine. Does there anyone who can point out the flaw in setup. Thanks and regards, Vijay Thakur

2 1

Can't initialize shared memory
by Uri Shacked 23 Jul '12

23 Jul '12

Hi, Anyone got this message " Can't initialize shared memory" ? When i run kamailio using "kamailio -m 512" it works OK When i run "/etc/init.d/kamailio start" i get the message above. my init script anf defaults are set with memory = 512. I played a little with 3.3.0 and now i am back to 3.2.4 can it be something to do with it? BR, Uri

2 1

Re: [SR-Users] How to rewritehostport() dynamically?
by Alex Balashov 20 Jul '12

20 Jul '12

Try modifying the request URI with $ru/$rU/$rd/$rp instead. See the pseudovariables page on the dokuwiki for more information. rewritehostport() is a legacy function that does not deal with variables. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/intel@intrans.baku.az wrote:Can I use rewritehostport() function with argument selected at time of call processing? I have a lua script, which gets address/port of next hop(from base) But how to call rewritehostport() with it? I can't find, how to call this function from lua script directly (I've tried sr.tm.rewritehostport() and rewritehostport() - no success) If i set pseudovariable (f.e $avp(relay)) and call rewritehostport("$avp(relay)") from kamailio.cfg later, i can see in logs such message: ERROR: tm [ut.h:327]: failed to resolve "$avp(relay)" Seems it use name of pv, not value? What should I do to call this function with correct parameter? -- _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

How to rewritehostport() dynamically?
by intel＠intrans.baku.az 20 Jul '12

20 Jul '12

Can I use rewritehostport() function with argument selected at time of call processing? I have a lua script, which gets address/port of next hop(from base) But how to call rewritehostport() with it? I can't find, how to call this function from lua script directly (I've tried sr.tm.rewritehostport() and rewritehostport() - no success) If i set pseudovariable (f.e $avp(relay)) and call rewritehostport("$avp(relay)") from kamailio.cfg later, i can see in logs such message: ERROR: tm [ut.h:327]: failed to resolve "$avp(relay)" Seems it use name of pv, not value? What should I do to call this function with correct parameter? --

1 0

Kamailio and IMS
by Apoorv Khare 20 Jul '12

20 Jul '12

Hi Everyone, I am trying to integrate IMS(*carstenbock/ims*) patch in Kamailio. But so far I could not find any proper documentation to load configuration files - pcscf, icscf, scscf, mgcf etc? Please let me know if there is anything available! Only relevant link I found - http://www.kamailio.org/w/2011/01/ims-extensions-available-for-testing/ AND this link does not seems to be update - http://www.ng-voice.com/our-solution/installation/ BR, Apoorv

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sr-users July 2012