June 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

[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

[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

Reply 503 was replaced to 500
by Uri Shacked 18 Jul '12

18 Jul '12

Hi, Kamailio server is behind our company's softswitch and acts as a sip application server. I notice that there are calls that the softswitch replied with 503 "service unavailable" and kamailio sent to the originator leg 500 "service unavaileable". When kamailio recieved 504 or 502 it sends them back as is. shouldn't it be the same with 503? It also does not have a "to tag" in the CDR. And the "to tag" in the 503 that was recieved is not equal to the 500 reply "to tag" kamailio sent back. any ideas? BR, Uri

2 6

Kamailio Unexpectedly Terminating
by Akan 12 Jul '12

12 Jul '12

I have 2 servers running Solaris and Kamailio 3.2.3 where on one Kamailio is terminating when it tries to save the location for a register request and the other is producing a core dump when processing an Option request. I have one server handling Register request while the other sip server forwards the register requests and handles the other requests. I have included the backtraces from the core dumps and the output from the log for the registrar server as well as the command that is causing kamailio to terminate: if (!save("location")) sl_reply_error(); 4(3364) ERROR: *** cfgtrace: c=[/opt/kamailio-3.2/etc/kamailio/kamailio.cfg] l=714 a=17 n=if 4(3364) ERROR: *** cfgtrace: c=[/opt/kamailio-3.2/etc/kamailio/kamailio.cfg] l=711 a=26 n=save 14(3374) : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 15 14(3374) DEBUG: <core> [tcp_main.c:3555]: DBG: handle_ser_child: dead child 4, pid 3364 (shutting down?) 14(3374) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (1003743d8, 15, 0, 0x0) fd_no=18 called 0(3360) ALERT: <core> [main.c:751]: child process 3364 exited by a signal 10 0(3360) ALERT: <core> [main.c:754]: core was not generated 0(3360) INFO: <core> [main.c:766]: INFO: terminating due to SIGCHLD 6(3366) INFO: <core> [main.c:817]: INFO: signal 15 received 1(3361) INFO: <core> [main.c:817]: INFO: signal 15 received 2(3362) INFO: <core> [main.c:817]: INFO: signal 15 received 3(3363) INFO: <core> [main.c:817]: INFO: signal 15 received 5(3365) INFO: <core> [main.c:817]: INFO: signal 15 received 7(3367) INFO: <core> [main.c:817]: INFO: signal 15 received 8(3368) INFO: <core> [main.c:817]: INFO: signal 15 received 9(3369) INFO: <core> [main.c:817]: INFO: signal 15 received 10(3370) INFO: <core> [main.c:817]: INFO: signal 15 received 11(3371) INFO: <core> [main.c:817]: INFO: signal 15 received 12(3372) INFO: <core> [main.c:817]: INFO: signal 15 received 13(3373) INFO: <core> [main.c:817]: INFO: signal 15 received 14(3374) INFO: <core> [main.c:817]: INFO: signal 15 received 0(3360) DEBUG: presence_xml [presence_xml.c:347]: start 0(3360) ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Permission denied (13) 0(3360) DEBUG: <core> [db_pool.c:102]: removing connection from the pool 0(3360) DEBUG: db_postgres [km_pg_con.c:122]: PQfinish(100842470) 0(3360) DEBUG: db_postgres [km_pg_con.c:126]: pkg_free(1004c1f30) 0(3360) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(3360) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table 0(3360) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores 0(3360) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists 0(3360) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done Thanks Nathaniel

4 38

Binding arbitrary TLS socket
by Andreas Granig 02 Jul '12

02 Jul '12

Hi, Got some problems when I tried this, not sure how this is intended to work: listen=tls:192.168.51.133:4343 When starting kamailio 3.3, I get the following error: ERROR: <core> [tcp_main.c:2915]: ERROR: tcp_init: bind(10, 0x7f5eea45db24, 16) on 192.168.51.133:4343 : Address already in use ERROR: tls [tls_init.c:314]: Error while initializing TCP part of TLS socket 192.168.51.133:4343 What works though is if I explicitely bind a TCP socket one port below the TLS socket, like this: listen=tcp:192.168.51.133:4342 listen=tls:192.168.51.133:4343 Is this how it's intended to work? Is there a specific reason for this behavior? Andreas

2 2

cfgutils lock / unlock + $sht() variables.
by Vitaliy Aleksandrov 02 Jul '12

02 Jul '12

Hi all, I use $sht() pseudo-variabled provided by htable module for caching various information about sip customers. I thought that htable has some locking mechanism inside it and today while i was reading core-cookbook i payed attention to one example with the lock/unlock functions and $sht(): lock("calls-to::10.10.10.10"); $sht(a=>calls-to::10.10.10.10) = $sht(a=>calls-to::10.10.10.10) + 1; unlock("calls-to::10.10.10.10"); According to that example an explicit locking mechanism should be used when i want to write something to a hash table. On the other hand there are no locks in htable documentation. So the question is when do the explicit locks should be used and for what purpose ?

2 1

Re: [SR-Users] how to talk across domains, my domain to say, iptel.org
by Alex Balashov 30 Jun '12

30 Jun '12

Note that from a security point of view, this is not necessarily a desirable trait, without appropriate programmatic restrictions and/or ACLs. -- Alex -- 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/Auro Tripathy <Auro.Tripathy(a)wdc.com> wrote:Thanks Klaus, non-local routing appears to work out-of-the-box. Thank you again for confirming what I observed. -Auro From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Friday, June 29, 2012 3:52 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Auro Tripathy Subject: Re: [SR-Users] how to talk across domains, my domain to say, iptel.org The default configuration supports that already. Usually inbound routing does not differ if the call comes from a local subscriber or from an external domain. Outbound routing is performed if the domain in the request URI (ruri) is non-local. regards Klaus Am 29.06.2012 19:46, schrieb Auro Tripathy: Hi All: I’m a newbie, excuse the simple (no pun intended) question. I installed Kamailio and I’m able to set it up so that all subscribers in the *same* domain can talk/discover-presence, etc. What do I do so that subscribers in other domains, for example the iptel.org domain, can talk to subscribers in my domain? -Auro _______________________________________________ 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 talk across domains, my domain to say, iptel.org
by Auro Tripathy 30 Jun '12

30 Jun '12

Hi All: I'm a newbie, excuse the simple (no pun intended) question. I installed Kamailio and I'm able to set it up so that all subscribers in the *same* domain can talk/discover-presence, etc. What do I do so that subscribers in other domains, for example the iptel.org domain, can talk to subscribers in my domain? -Auro

2 2

Serctl
by Angel Valdes 29 Jun '12

29 Jun '12

Please is any variable available to stop screen scrolling when you use serctl showdb and you have that many users list bigger than your screen and you need to see the top part of the list Thanks Angel Valdes IT Manager 201-920-2908 Talent is a gift that brings with it an obligation to serve the world, and not ourselves, for it is not of our making. Jose Marti

2 1

Re: [SR-Users] Nathelper module, FLT_NATS, FLT_NATB
by Richard Brady 28 Jun '12

28 Jun '12

Klaus / Daniel Thanks again for assistance with this. I've tried the solution based on add_contact_alias() and handle_ruri_alias() and it works perfectly. Richard On 22 June 2012 13:47, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > > > On 22.06.2012 13:50, Richard Brady wrote: >> >> Thanks guys, fantastic answers. >> >> You mention that NAT detection happens before save() and the flag is set >> by lookup() which makes much more sense. However, if Kamailio is not the >> registrar, as is the case with my current project, those functions are >> not called, so an alternative is needed. There are clearly several >> options. >> >> The solution I have gone for is to replace fix_nated_register() with >> fix_nated_contact() so that the REGISTER request is relayed with a >> modified Contact header containing the external ip:port of the client. > > > The cleanest solution would be to use add_contact_alias() and > handle_ruri_alias(). The do not change the contact but put the public > address into a uri parameter. Thus, the URI seen by the client is always the > one it sends: > http://www.kamailio.org/docs/modules/3.2.x/modules_k/nathelper.html#id25504… > > >> That is then stored by the registrar (FreeSWITCH in my case) and used >> later to originate calls for that user. The FreeSWITCH know to send >> those calls to Kamailio through either use of the Path header and module >> in Kamailio, or through static configuration of fs_path or proxy >> parameters in FreeSWITCH. > > > This is fine. > > >> The works for the first INVITE to the registered client behind NAT. But >> that client sends back a 200 OK with a Contact header containing its >> private IP address, and so fix_nated_contact() needs to be invoked on >> that response, and normally it would be due to FLB_NATB being set, but >> if Kamailio was not the registrar then that flag is not set. So I need >> to detect NAT on the client at the time of receiving the reply, or >> alternatively by having the registrar store a cookie and setting it >> based on that. > > > You are correct. For in-dialog messages received from SIP clients I would > always use add_contact_alias() and remove the NAT flags completely. > >> >> I suppose then my next question then is can I call nat_uac_test() on a >> UAS? > > > Today, almost any SIP clients are SIP symmetric. This means, that they > receive SIP from the some port/connection where they send. Thus, skip the > NAT tests completely and always use add_contact_alias() for messages receive > from SIP clients and handle_ruri_alias() for messages sent to SIP clients. > > Depending on if you use Freeswitch also as media relay you can also remove > the media proxy stuff from the Kamailio config. > > On important thing is NAT-keep-alive. This is usually done by nathelper > module by querying the location table for contact with NAT-flag set. Thus, > either you do NAT keep-alive by freeswitch (e.g sending OPTIONS requests) or > do it in your Kamailio proxy (e.g. set the nat_bflag > http://www.kamailio.org/docs/modules/3.2.x/modules_k/usrloc.html#id2541477 > and call save("location","0x02") before relaying REGISTER to freeswitch. > Then Kamailio will do NAT-pinging. Note, if you want to keep-alive only for > successfully registered clients, you man want to call save() in the > reply-route of the REGISTER request). > > regards > Klaus

4 5

Git Error on Kamailio-3.3 Install
by SamyGo 28 Jun '12

28 Jun '12

Hello, I'm setting up new Kamailio 3.3 from git ref-link: http://www.kamailio.org/wiki/install/3.3.x/git I'm using CentOS 5.8 and when I execute the line # git checkout -b 3.3 origin/3.3 it gives me error. fatal: git checkout: updating paths is incompatible with switching branches. Did you intend to checkout 'origin/3.3' which can not be resolved as commit? Can I skip it ! o need to do something about it !! Regards, Sammy G.

3 6

Segfault running 3.3.0 with registrar/p_usrloc/path
by Dan-Cristian Bogos 28 Jun '12

28 Jun '12

Hey Guys, After upgrading from 3.2.x to 3.3.0 (out of debian packages and fixing db side) Kamailio is crashing on processing REGISTER. This crash only happens for updates (or if a stale contact for the same AOR is already present), not initial REGISTERs. Attached you can find the log taken with debug=6 as well as the REGISTER package producing the segfault. Kamailio version: """ root@PrxyDev:/etc/slocator# kamailio -V version: kamailio 3.3.0 (i386/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, F_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 22:17:07 Jun 18 2012 with gcc 4.4.5 """ Syslog msg in debug=2: """ Jun 27 15:46:22 PrxyDev kernel: [11330.002854] kamailio[6959]: segfault at 0 ip (null) sp bf8d531c error 4 in kamailio[8048000+1ea000] Jun 27 15:46:22 PrxyDev /usr/sbin/kamailio[6970]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 10 Jun 27 15:46:22 PrxyDev /usr/sbin/kamailio[6957]: ALERT: <core> [main.c:785]: child process 6959 exited by a signal 11 Jun 27 15:46:22 PrxyDev /usr/sbin/kamailio[6957]: ALERT: <core> [main.c:788]: core was not generated """ Thanks in advance for any tip! DanB

3 8

Re: [SR-Users] TM module timer question
by Gertjan Wolzak 28 Jun '12

28 Jun '12

Thanks guys for the info. Will try again and keep on it. Rgds, Gertjan On 6/28/12 08:55:27 AM Daniel-Constantin Mierla wrote; Hello, On 6/27/12 6:08 PM, Corey Edwards wrote: > On 06/26/2012 08:36 AM, Gertjan Wolzak wrote: >> Which timer do I need to use to get a failure route to reroute the >> call to a pstn gateway? >> >> >> >> I have tried the fr_timer which works, I set it to 5000, downside is >> that when I am in wifi range, I only have 5 seconds to answer the call. >> >> >> >> The fr_inv_timer is not the right one I think, as I will not get any >> message from the called number, as that device is not connected?. > I'm not an expert on this, but I'll post my experience and I hope that > helps. I'm using Kamailio 1.5, so perhaps YMMV. > > I have fr_timer set to 2 seconds and fr_inv_timer set to 60 seconds. > The latter can be overridden by fr_inv_timer_avp so users can alter > their own timers. I do not let users change fr_timer. > > The behavior I see is the device has 2 seconds to give a provisional > response to an INVITE. If the device has powered off, changed IP, etc. > this timer will quickly return an error. > > If the device is responsive and, for example, sends a 180 Ringing, the > fr_timer ceases to apply and the fr_inv_timer waits 60 seconds before > giving up and triggering a 408 Timeout. it is the same in 3.x, these two parameters are the right to be used in this way. Have in mind that in 3.x the accuracy is in milliseconds: <http://kamailio.org/docs/modules/stable/modules/tm.html#fr_timer> http://kamailio.org/docs/modules/stable/modules/tm.html#fr_timer Another new features in 3.x is that apart of using avps to set it per transaction (still in seconds), there is a new function that can be used to set it in milliseconds per transaction: <http://kamailio.org/docs/modules/stable/modules/tm.html#t_set_fr> http://kamailio.org/docs/modules/stable/modules/tm.html#t_set_fr Cheers, Daniel

1 0

system call
by Kelvin Chua 28 Jun '12

28 Jun '12

hi guys, i'm currently using cron to alert the admin via email of any irregularities (e.g. pike) but i'm wondering, is it possible to do a system call from the cfg script?

2 1

Question about git update
by Ricardo Martinez 28 Jun '12

28 Jun '12

Hello. Can someone help me on how to update a module patch using GIT? I need to update the module dialog….. Currently I have the directory /usr/local/src/kamailio-3.2 And under this I have kamailio/ all this was created using : git clone --depth 1 git://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 3.2 origin/3.2 I’ll appreciate any help. Thanks in advance, Ricardo Martinez.-

2 1

Questions about login to account
by Daniel Butenko 28 Jun '12

28 Jun '12

3 2

Re: [SR-Users] TM module timer question
by Corey Edwards 28 Jun '12

28 Jun '12

On 06/26/2012 08:36 AM, Gertjan Wolzak wrote: > Which timer do I need to use to get a failure route to reroute the call > to a pstn gateway? > > > > I have tried the fr_timer which works, I set it to 5000, downside is > that when I am in wifi range, I only have 5 seconds to answer the call. > > > > The fr_inv_timer is not the right one I think, as I will not get any > message from the called number, as that device is not connected…. I'm not an expert on this, but I'll post my experience and I hope that helps. I'm using Kamailio 1.5, so perhaps YMMV. I have fr_timer set to 2 seconds and fr_inv_timer set to 60 seconds. The latter can be overridden by fr_inv_timer_avp so users can alter their own timers. I do not let users change fr_timer. The behavior I see is the device has 2 seconds to give a provisional response to an INVITE. If the device has powered off, changed IP, etc. this timer will quickly return an error. If the device is responsive and, for example, sends a 180 Ringing, the fr_timer ceases to apply and the fr_inv_timer waits 60 seconds before giving up and triggering a 408 Timeout. Corey

2 1

A question about installing kamalio
by Aurel Sandu 28 Jun '12

28 Jun '12

Hello all of you, I tried to install kamallio 3.3 version released a couple days ago . I followed the steps mentioned here https://www.kamailio.org/wiki/install/3.3.x/git Everything works fine until I have to create mysql database . Everything goes fine until I have to install mysql database (step 7) : I run "sudo /usr/local/sbin/kamdbctl create" for database creation but I get this : INFO: test server charset /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 105: mysql: command not found /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 106: mysql: command not found 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 ... /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 71: mysql: command not found ERROR: Creating core database and grant privileges failed! It seems that mysql -serverside- is not installed (my opinion ...) .What do you think? Many Thanks, Aurel ps: please excuse me if this is not right forum for this question but I did not find the right one ....

2 1

Re: [SR-Users] dlg_refer
by Alex Balashov 28 Jun '12

28 Jun '12

It sends an actual REFER. -- Alex -- 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/Uri Shacked <ushacked(a)gmail.com> wrote:Hi, before trying dlg_refer, does anyone know, does it send REFER to the caller/callee? or doeas it do reinvite? i ask this because my class 4 softswitch does not support REFER.... thanks, Uri

1 0

dlg_refer
by Uri Shacked 28 Jun '12

28 Jun '12

Hi, before trying dlg_refer, does anyone know, does it send REFER to the caller/callee? or doeas it do reinvite? i ask this because my class 4 softswitch does not support REFER.... thanks, Uri

1 0

Subscription-State: terminated;reason=terminated?
by Min Wang 27 Jun '12

27 Jun '12

HI when I removed 102 from 101's contact list (using jitsi nightly 1.1 build), kamailio 3.3 send out NOTIFY to 102 like this: NOTIFY sip:102@192.168.122.147:5060;transport=udp;registering_acc=192_168_122_32 SIP/2.0. Via: SIP/2.0/UDP 192.168.122.32;branch=z9hG4bK1bfb.afbf0a85.0. To: sip:102@192.168.122.32;tag=f6a40771. From: sip:101@192.168.122.32;tag=a6a1c5f60faecf035a1ae5b6e96e979a-5724. CSeq: 4 NOTIFY. Call-ID: c7c52dd058268596ec84dd3c645a2f17(a)0.0.0.0. Content-Length: 0. User-Agent: kamailio (3.3.0-rc0 (x86_64/linux)). Max-Forwards: 70. Event: presence. Contact: <sip:192.168.122.32:5060;transport=udp>. Subscription-State: terminated;reason=terminated. <----------------- Note the reason code is:terminated. From rfc3265, The defined reason codes are: deactivated/ probation/rejected/ timeout/giveup/noresource What is the reason to send: reason=terminated instead one of the well-defined reason codes? There was a discussion regarding at: http://sip-router.org/tracker/index.php?do=details&task_id=133 <http://sip-router.org/tracker/index.php?do=details&task_id=133> but I did not see the explaination of reason=terminated. Thanks min

3 13

default kamailio configuration
by Vitaliy Aleksandrov 27 Jun '12

27 Jun '12

Hi All, After reading default kamailio configuration i can't understand why does kamailio remove preloaded route headers from the incoming initial INVITE before calling record_route(). remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); What will happen if an INVITE comes from another SIP proxy which wants to stay in the middle of the dialogue ? For example, during Re-INVITE UA from our local site will construct in-dialog INVITE with R-URI taken from the remote contact and send it to our local proxy according to RR. Local proxy will process it in route[WITHNDLG] where loose_route() function will remove top most RR header because it points to us and since there is no more RR records forward the request to a destination from the R-URI. As i understand in that case remote proxy will not receive any in-dialog requests from us.

3 3

kamailio.org server maintenance - estimated 1 hour downtime, 26/27 June
by Daniel-Constantin Mierla 27 Jun '12

27 Jun '12

Hello, during the night of 26/27th of June, Central European Standard Time (GMT+1, with daylight saving time on), the server hosting kamailio.org and lists.sip-router.org will be down for approx 1 hour. This is a planned maintenance work to upgrade the server to a new infrastructure. Hopefully everything goes smooth and gets back online quickly, the work should start about 2:00am on the 27th of June, but don't get nervous if takes longer or happens at different time. Among the most important affected services: - the mailing lists (all mailed in this message) - main website and kamailio's wiki systems - download folders - documentations (for modules and other html tutorials) The sip-router.org server will NOT be affected, so next services will be available: - sip-router.org web site and its wiki - GIT repository - issue tracker system If things are not going as expected, tomorrow we will post news about on this server, at http://sip-router.org This is a good opportunity to thank again to voztele.com for sponsoring the server and taking care of the maintenance work. 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

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Re: [SR-Users] Alias possible for presence...
by Daniel-Constantin Mierla 26 Jun '12

26 Jun '12

Hello, On 6/20/12 9:19 AM, Gertjan Wolzak wrote: > > Well, the account name is not public, but I would like to have the > possibility to have a "private" account for registering and a > "public"account to publish. > > The goal is to have others to be able to send messages to your > "public" id, while you log in with your "private" id. > most of user agent have two fields, one to be used as SIP id and another one for authentication id. > I know, think the next step is to get an communicator that has that > possibility..... > > So I want to be able to publish the "public" account and not the > "private" one. > > For example: > > I register to the presence server with uid gertjan and my password, > but I want to be known to others as newbie. > > So my friend can send an invitation/message to newbie as well as to > gertjan . > > Just like the db_alias which is used in the kamalio for linking sip > accounts to aliases. > For presence, people have to subscribe to a SIP id, in this case it your public ID known by everybody. The easiest will be that the user agent will publish in behalf of the public id. But ultimately, you can replace the values in headers and xml body (see textops/xmlops functions), then do presence handling, after applying the changes to the message (see textopsx module). Cheers, Daniel > Rgds, > > Gertjan > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* dinsdag 19 juni 2012 10:27 > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* Gertjan Wolzak > *Subject:* Re: [SR-Users] Alias possible for presence... > > Hello, > > On 6/14/12 10:28 AM, Gertjan Wolzak wrote: > > Hello All, > > I have the following challenge. > > We are using kamailio 3.2 and have enabled presence. Which works fine. > > But we would like to be able to work with aliases, so that the > account name and "presentation" name can be different. > > Any Ideas... Or did I miss the documentation on that? > > the account name is public or not? If it is some internal id, > eventually is used for authentication, but for presence the contacts > should know the public id. > > Or maybe you can give some example to understand how you use aliases > in the context of presence... > > Cheers, > Daniel > > > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/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 > > > -- > This message was scanned by Foize and is believed to be clean. > Click here to report this message as spam. > <http://filter.foize.com/cgi-bin/learn-msg.cgi?id=> > > > > _______________________________________________ > 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 -- 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

2 2

Kamailio : Listening on multiple ports behind NAT
by Reda Aouad 26 Jun '12

26 Jun '12

Hello, I am trying to implement the following configuration : - Kamailio as a SIP proxy/registrar behind a one-to-one NAT (port number is not modified) listening on ports 5060 and 53 (and more ports in the future) - aliases correctly configured : alias= udp: public_ip:53 alias= udp: public_ip:5060 alias= udp: hostname:53 alias= udp: hostname:5060 - listen directive correctly on private ip address and both ports : listen=udp:private_ip:53 listen=udp:private_ip:5060 - advertised_address=public_ip - record_route_preset("public_ip") is used to announce the public IP address in the RR header - user A : registered on port 5060 - user B : registered on port 53 Suppose user A tries to call user B. The Record-Route header in the INVITE forwarded from Kamailio to user B should contain the port number on which user B is connected (53), to force user B to send future requests to that port number. But I have no method to know which port user B is connected to, and that problem is aggravated when user B has multiple registrations on different port numbers and parralel forking is done. Declaring advertised_port doesn't solve the problem. I cannot force port number 53 in record_route_preset("public_ip:53") since it wouldn't work when user B calls user A. Using the record_route( ) function, Kamailio doesn't use the advertised_address to construct the RR header. Another problem is that the record_route_preset function clears the DID cookie set by the dialog module, which makes Kamailio fallback to SIP elements to match the request to an existing dialog, thus dialog matching becomes slower, and performance is an issue for me. Any suggestions? I know that one solution would be to run Kamailio with a public IP address and no NAT, but unfortunately it's not possible. I suggest that the function record_route( ) takes a public IP address as a parameter, still doing what it does (correct record routing and cookie addition did=xxx and loose route lr=on), but only replacing the private IP address on which Kamailio listens with a public IP address. Or that the record_route( ) function uses the advertised_address to construct the RR header. Thank you RA

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Kamailio v3.3.0 packages
by Daniel-Constantin Mierla 25 Jun '12

25 Jun '12

Hello, I'm summarizing the status of packages for latest release v3.3.0. Debian/Ubuntu packages are ready in the usual APT repository, including nightly builds of the GIT branch 3.3, details at: * http://www.kamailio.org/wiki/packages/debs RPMs for CentOS/RedHat, Fedora, openSuse (various versions for each OS) are now available on opensuse build factory repository, details at: * http://www.kamailio.org/wiki/packages/rpms There you find the note about Peter Dunkley's repository, which has also builds for Raspberry Pi. 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 0

Over 100% of CPU usage on kamailio
by Konstantin M. 25 Jun '12

25 Jun '12

Hello, Is there any reason to consume over 135% of CPU usage on kamailio ? (sreenshot: http://i48.tinypic.com/2u72hr8.png). Any ideas ? Thanks!

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Fwd: Can Kamailio be used to redirect media between a client that switches from wifi to 3g/gsm
by Shaun Clark 25 Jun '12

25 Jun '12

Forgot to post the response to the list as well. Date: Fri, Jun 22, 2012 at 6:57 AM Subject: Re: [SR-Users] Can Kamailio be used to redirect media between a client that switches from wifi to 3g/gsm To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Thanks for the response! I see a series of what I believe are re-REGISTER statements: Message sent: (to dest=75.101.244.XXX:5060) REGISTER sip:75.101.244.XXX SIP/2.0 Via: SIP/2.0/UDP 10.165.27.161:2407;rport;branch=z9hG4bK1839704852 From: <sip:990XX@75.101.244.XXX>;tag=1689684502 To: <sip:990XX@75.101.244.XXX> Call-ID: 1867622191 CSeq: 1 REGISTER Contact: <sip:990XX@10.165.27.161:2407;line=daeb0d9351eff22> Max-Forwards: 70 User-Agent: Linphone/3.4.0 (eXosip2/unknown) Expires: 3600 Content-Length: 0 Received message: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.165.27.161:2407;received=32.158.143.61 ;rport=2407;branch=z9hG4bK1839704852 From: <sip:990XX@75.101.244.XXX>;tag=1689684502 To: <sip:990XX@75.101.244.XXX>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.e10d Call-ID: 1867622191 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="75.101.244.XXX", nonce="4fe476e500000e00aa8700d49b8c668f4c6f1e6a367f2XXX" Server: Kamailio Content-Length: 0 REGISTER sip:75.101.244.XXX SIP/2.0 Via: SIP/2.0/UDP 10.165.27.161:2407;rport;branch=z9hG4bK123406454 From: <sip:990XX@75.101.244.XXX>;tag=1689684502 To: <sip:990XX@75.101.244.XXX> Call-ID: 1867622191 CSeq: 2 REGISTER Contact: <sip:990XX@10.165.27.161:2407;line=daeb0d9351eff22> Authorization: Digest username="990XX", realm="75.101.244.XXX", nonce="4fe476e500000e00aa8700d49b8c668f4c6f1e6a367f2XXX", uri="sip:75.101.244.XXX", response="1e1d558894f2c05c322c76efbb2f9XXX", algorithm=MD5 Max-Forwards: 70 User-Agent: Linphone/3.4.0 (eXosip2/unknown) Expires: 3600 Content-Length: 0 Received message: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.165.27.161:2407;received=32.158.143.61 ;rport=2407;branch=z9hG4bK123406454 From: <sip:990XX@75.101.244.XXX>;tag=1689684502 To: <sip:990XX@75.101.244.XXX>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.8b5a Call-ID: 1867622191 CSeq: 2 REGISTER Contact: <sip:990XX@10.165.27.161:2407;line=daeb0d9351eff22>;expires=120, <sip:990XX@50.43.101.83:51879;line=59ecc207f06f4e9>;expires=81 Server: Kamailio Content-Length: 0 But after this I would expect to see an INVITE but one is never sent, but if I switch back to the original IP on that device the call is reconnected, so it proves we're missing an INVITE I believe. What do I need to do on the server side to force a re-INVITE to be sent after this registration occurs? Thanks! On Fri, Jun 22, 2012 at 1:14 AM, Klaus Darilion < klaus.mailinglists(a)pernau.at> wrote: > Hi Shaun! > > Your problem description is too short to give you any good help. > > Use tcpdump (or other tools) to capture the scenario with Asterisk and > Kamailio. Then compare them to find out why it doesn't work. > > Is media sent directly to Asterisk then it ca not be the problem of > Kamailio. > > I hope the mobile client is smart enough to also send a reINVITE when > getting the new IP address (of the mobile connection) with proper Contact > header - otherwise it can not receive SIP requests from Asterisk. > > regards > Klaus > > > On 20.06.2012 18:07, Shaun Clark wrote: > >> The use case is that I have a SIP client registered to Kamailio talking >> to an Asterisk box connected to the PSTN. The client is a mobile phone >> and the user is connected to wifi. The user then steps out of wifi range >> and the phone drops the connection and picks up the 3g data connection. >> I want the media stream to reconnect to the client and the call to >> resume without having to redial. This works now if the client is >> directly connected to the Asterisk machine, but not when I am routing >> through my Kamailio server. How do I go about this, examples are always >> appreciated, thanks! >> >> >> ______________________________**_________________ >> 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<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >> >> > > --

2 1

ERROR: uri2dst: bad_uri
by Barthel Marco (CI/AFU1) 25 Jun '12

25 Jun '12

Hi, we are using kamailio at Bosch with some several thousand users. From time to time we encounter the following in the logs of kamailio: /usr/sbin/kamailio[9247]: ERROR: tm [ut.h:255]: ERROR: uri2dst: bad_uri: sip: /usr/sbin/kamailio[9247]: ERROR: tm [t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches Am I right that this means a relay/forward happened and either $ru or $du was just set to "sip:" ? Thanks in advance. Mit freundlichen Grüßen / Best regards Marco Barthel Robert Bosch GmbH (CI/AFU1) www.bosch.com Tel. +49 711 811-3602341 marco.barthel(a)de.bosch.com Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart, HRB 14000; Aufsichtsratsvorsitzender: Hermann Scholl; Geschäftsführung: Franz Fehrenbach, Siegfried Dais; Stefan Asenkerschbaumer, Bernd Bohr, Rudolf Colm, Volkmar Denner, Christoph Kübel, Uwe Raschke, Wolf-Henning Scheider, Werner Struth, Peter Tyroller

2 1

Nathelper module, FLT_NATS, FLT_NATB
by Richard Brady 22 Jun '12

22 Jun '12

Hi folks I've configured Kamailio as a simply multi-homed proxy offering NAT traversa, TCP connection management, etc to FreeSWICH behind it. FreeSWITCH acts as the registrar. I'm trying to understand the default config and the following things don't make sense to me (I can't find documentation on these): 1. The nat=yes flag. What does it mean exactly and why is it on the RR and not on the Contact/R-URI? Also, when double RR is enabled, why does it get added to both headers? 2. The FLT_NATS and FLB_NATB flags. I'd like to know what this stands for, I guess "flag for transaction - NAT" and "flag for branch - NAT" but what do the S and the B stand for? More importantly what do these flags mean? Is it correct to say that the first indicates the UAC / A-leg / requester is behind NAT and the second that the UAS / B-leg / responder is behind NAT? 3. In the the NATMANAGE route of the default config we have: if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } Which to me means "if the message is a request inside an existing dialog ..." and I'm thinking, isn't this a bit late in the dialog to be setting NAT flags? I would expect to set the flag for the A-leg on the incoming request, and the flag for the B-leg on the incoming reply. I can't point in waiting for the ACK to try and work out whether this leg is behind NAT. And also it seem this set of if statements will also match incoming requests from the A-leg, so why are we setting a brach flag? Does that leg also also get the properties of a branch? 4. When I get a 200 OK coming from behind NAT it is not subjected to fix_nated_contact(), and this seems to be because it doesn't have the FLB_NATB flag set: if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } Any help clarifying would be much appreciated. Thanks, Richard -- Richard Brady

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Can Kamailio be used to redirect media between a client that switches from wifi to 3g/gsm
by Shaun Clark 22 Jun '12

22 Jun '12

The use case is that I have a SIP client registered to Kamailio talking to an Asterisk box connected to the PSTN. The client is a mobile phone and the user is connected to wifi. The user then steps out of wifi range and the phone drops the connection and picks up the 3g data connection. I want the media stream to reconnect to the client and the call to resume without having to redial. This works now if the client is directly connected to the Asterisk machine, but not when I am routing through my Kamailio server. How do I go about this, examples are always appreciated, thanks!

2 1

Geographically Distributed Active Sites/Servers
by SamyGo 22 Jun '12

22 Jun '12

Hello, I've been pondering over an architecture of distributed VoIP services on different geographic regions. I'm sure its nothing new and there are lots of guys have ideas to share. There are multiple SIP enabled Regions/locations, each has its own media services and full services capability. Now Imagine I've unique SIP users registering on each Region. What I want to achieve is as follows: 1- User A from Region-A dials User-B registering on Region-C 2- Region-A inquires all Regional-Nodes and in response only one Regional Node (Region-C) raises hand that it has this particular user registered. 3- Upon this reply Region-A Kamailio relays the call to Region-C kamailio and we get a successful call. The SIP users are mobile and can register to different regions depending upon their location so thats why I need step-2. Offline SIP user's will be handled by the destination number area code and routed to the closest serving VoIP region. How can we use the new modules like DMQ or Presence to achieve this !1 Regards, Sammy Go.

1 0

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Charles Chance 22 Jun '12

22 Jun '12

Hi Daniel It is fixed :-) What was the problem? Thanks, Charles Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Hello, can you try with latest master branch -- it should be fixed by: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=1d89d7b… Cheers, Daniel On 5/29/12 5:52 PM, Daniel-Constantin Mierla wrote: Hello, finally had a chance to look over it. I guess it is some race, resulting in a memory space being overwritten. The 200ok for publish with ringing state is processed more or less at the time when 200ok for invite comes and dialog changes state. Can you re-compile with MEMDBG=1 (edit the Makefile.defs or give it in the command line parameter) and try again? Maybe the memory manager can catch the fault operation and provide more details. Cheers, Daniel On 5/17/12 6:58 PM, Charles Chance wrote: Hi Daniel, I have sent the messages over to you. I look forward to hear what you can find. Best, Charles From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: 17 May 2012 15:06 To: Charles Chance Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List' Subject: Re: FW: (Devel) Segmentation fault using pua_dialoginfo Hello, the content of ps->param is pretty much all invalid. So looks like a callback on a freed/invalid parameter, could be a double-callback execution. Set debug=3 and send all the messages for such case -- it might be quite a lot of them, so send them to my email address, the mailing list has a limit of message size. Cheers, Daniel On 5/17/12 2:13 PM, Charles Chance wrote: Hi, Requested output as follows: (gdb) frame 1 #1 publ_cback_func (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) at send_publish.c:246 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); (gdb) p *ps $2 = {req = 0x0, rpl = 0xb7c30340, param = 0xb3dc3ff4, code = 412, flags = 0, branch = 0, t_rbuf = 0x0, dst = 0x0, send_buf = {s = 0x0, len = 0}} (gdb) p *(ua_pres_t*)(*ps->param) $3 = {id = {s = 0x20455942 <Address 0x20455942 out of bounds>, len = 980445555}, pres_uri = 0x40323531, event = 875444279, expires = 875703856, desired_expires = 858861105, flag = 808794676, db_flag = 1394618421, cb_param = 0x322f5049, next = 0xa0d302e, ua_flag = 979462486, etag = { s = 0x50495320 <Address 0x50495320 out of bounds>, len = 808333871}, tuple_id = {s = 0x5044552f <Address 0x5044552f out of bounds>, len = 775436064}, body = 0x322e3034, content_type = { s = 0x312e3134 <Address 0x312e3134 out of bounds>, len = 1648047155}, watcher_uri = 0x636e6172, call_id = { s = 0x397a3d68 <Address 0x397a3d68 out of bounds>, len = 1647593320}, to_tag = {s = 0x3435634b <Address 0x3435634b out of bounds>, len = 842149473}, from_tag = { s = 0x63306461 <Address 0x63306461 out of bounds>, len = 808334648}, cseq = 1767246349, version = 1394621025, outbound_proxy = 0x322f5049, extra_headers = 0x552f302e, record_route = { s = 0x31205044 <Address 0x31205044 out of bounds>, len = 825111097}, remote_contact = {s = 0x312e3836 <Address 0x312e3836 out of bounds>, len = 976498224}, contact = { s = 0x36373135 <Address 0x36373135 out of bounds>, len = 1701985073}} Yes, it is a test server so very happy to arrange for remote access if required. In the meantime, I will do a little more digging to try to find out why entity is null. Cheers, Charles -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2176 / Virus Database: 2425/5004 - Release Date: 05/16/12 -- 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 -- 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

2 1

New developer - Vicente Hernando
by Daniel-Constantin Mierla 21 Jun '12

21 Jun '12

Hello, I am introducing Vicente as a new registered developer - he submitted lately very useful patches to ndb_redis module (e.g., array support in replies, redis free function for config), new ones being on the pipe -- he can present better his plans or you just watch the commits. Mainly he will be in charge with ndb_redis module, being a heavy user of it, but there have been submissions from him to other parts of code, we welcome improvements and new good features anywhere! Welcome and looking forward to your contributions! 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

7 6

$200 bounty
by copycall 21 Jun '12

21 Jun '12

hello, install and configure kamailio on a godaddy load-balanced cloud server and create a vpn with the customer site. the objective is to use the customer's sip provider, which requires a static ip address, with a dhcp connection. please contact me off-list if you are interested. dave(a)copycall.com thank you, dave

10 25

Bug with server originated call
by David 20 Jun '12

20 Jun '12

Hello, I am using the example that came with my kamailio deb file to launch a call server side. I send a dummy INVITE to the phone, when the phone responds 200 OK, I reply with a REFER to send the phone to the desired number. ( ctd.sh.gz ) My problem is that if I don't answer the ctd call, the phone rings for a long time ( or endlessly on certain devices ). I tried adding an Expires: 10, and it did not make any difference. Is there a way to cancel this call from the ctd script ? David

2 2

Kamailio v3.3.0 Released - new major version is out
by Daniel-Constantin Mierla 20 Jun '12

20 Jun '12

Kamailio v3.3.0 is out – a major release with a very large number of new features and improvements. This release is a result of about 6 months of development and 2 months of testing from the teams of Kamailio (OpenSER) and SIP Express Router (SER) projects. Version 3.3.0 comes with 7 brand new modules in addition to a lot of fresh features in core and old modules. You can read detailed release notes at: * http://www.kamailio.org/w/kamailio-v3-3-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy SIP routing in a secure, flexible and easier way with Kamailio v3.3.0! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Re: [SR-Users] simple test case: add user/delete user on kamailio 3.3 presence server
by Daniel-Constantin Mierla 20 Jun '12

20 Jun '12

Hello, On 6/19/12 4:49 PM, Min Wang wrote: > Hi > > Here is another similar case which does not work. maybe it is related > the above issue as well. > > Jitsi 1.1 client: 101/103 , with xcap as storage > proxy: kamailio 3.3 > ip network is: 192.168.122.0 > > the setup is like the following: > > 101(.224) --- proxy (.32) --- 103 (.1) > > After all presence related db are truncated, kamailio was restart. > then bring 101,103 online. > > On 101 add 103, 103 prompt up authorization window, click ok. > now on 103, I can see 101 online status > > but on 101, I can not see 103 status <----the first trace file: > the jitsi-101-add-103-confirm-2.pcap > > (it is grey, even manually change 103 status from online to away > etc, on 101, the 103 always showed grey. <--- second trace file: > jitsi-101-103-change.pcap) > > Please see the attached pcap files. > > > the watcher table shows: > > id presentity_uri watcher_username watcher_domain > event status reason inserted_time > 1 sip:103@192.168.122.32 101 192.168.122.32 presence > 2 1340114579 > 2 sip:101@192.168.122.32 103 192.168.122.32 presence > 1 1340114582 the first record above shows that the status is 2, which means 'pending'. The subscription of 101 was not accepted yet by presentity 103. For the second record, status is 1, which means active subscription. 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

2 3

simple test case: add user/delete user on kamailio 3.3 presence server
by Min Wang 20 Jun '12

20 Jun '12

4 7

rtimer question
by Alex Balashov 20 Jun '12

20 Jun '12

Hi, I use 'rtimer' and 'mqueue' to asynchronously defer certain heavy DB tasks on which call processing does not depend. I was hoping to get a clarification on the exact behaviour of 'rtimer' when tasks don't complete as fast as they should. Let's say that I set the rtimer route interval to half a second (500 ms), as afforded by new Kamailio 3.3.0 capabilities. What happens if the tasks I am running inside the rtimer route don't complete in that interval? Is that rtimer thread forceably terminated, or is it allowed to run as long as necessary to complete the task? Is there an absolute timeout? If the latter, I assume that another, parallel rtimer route will be started in the background after 500 ms. I am given to understand that there is a thread pool. How big is it? Can I configure its size anywhere? Is there a practical limit? Finally, is there 100% assurance that 'mqueue' is safe to consume for multiple rtimer processes, possibly ones tripping over each other due to unpredictable delays? Or is it designed only to provide blocking mutex for data passed from a SIP worker thread to a different kind of thread? 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/

2 1

kamailio core request dropping - no reply
by Uri Shacked 20 Jun '12

20 Jun '12

Hi, I am testing kamailio replies when an INVITE or another request arrives with lets say, "VIA" header missing.... The core drops the request. But, there is no reply for the originator (so it keep on resending the request...) Why? Uri

4 7

Siremis routing support does not include drouting
by Yufei Tao 20 Jun '12

20 Jun '12

Hello Seems Siremis 3.2.1 doesn't support configuration of drouting. Any plans that it will in the future? I hope to use it to make my drouting configuration easier. What I want to do is to route prefixed numbers to external number at external domains, depending on the prefix (numbers+alphabets+dot and maybe+underscore+hyphen as well). Maybe I should use PDT module instead as it is supported by Siremis? Thanks! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

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Re: [SR-Users] Alias possible for presence...
by Daniel-Constantin Mierla 20 Jun '12

20 Jun '12

Hello, On 6/14/12 10:28 AM, Gertjan Wolzak wrote: > > Hello All, > > I have the following challenge. > > We are using kamailio 3.2 and have enabled presence. Which works fine. > > But we would like to be able to work with aliases, so that the account > name and "presentation" name can be different. > > Any Ideas... Or did I miss the documentation on that? > the account name is public or not? If it is some internal id, eventually is used for authentication, but for presence the contacts should know the public id. Or maybe you can give some example to understand how you use aliases in the context of presence... 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

2 1

Re: [SR-Users] simple test case: add user/delete user on kamailio 3.3 presence server
by Min Wang 19 Jun '12

19 Jun '12

HI Since my previous email has attachment , it is still waiting for the approval. So here is again the text part ( please see the email below) According to the trace, here is the call flow for this case: 101(.224) --- proxy (.32) --- 103 (.1) 101/103 are jitsi 1.1 nightly build client, the configure is to use SIMPLE/xcap on clients. the call flow for 101 adding 103 as contact 101 proxy 103 | | | user-add-103 | | | | | | xcap put pres/rls | | |--------------------------> | | | | | | SUBSCRIBE (presence/103) | | |---------------------------> | | | 202 ok | | |<----------------------------| | | NOTIFY( presenc/pending) | | |<----------------------------| | | | NOTIFY ( winfo/101 pending) | | |---------------------------------->| | 200 ok | | |--------------------------> | | | | 200 OK | | |<----------------------------------| | | auth window pop | | user accept | | SUBSRIBE(presence/101) | | |<----------------------------------| | | 202 OK | | |---------------------------------->| | NOTIFY (winfo,103 active) | | |<-------------------------- | | | 200 ok | | |--------------------------> | | | | NOTIFY (presence/101) | | |---------------------------------> | | | 200 OK | | |<----------------------------------| | NOTIFY (presence/103 open) | | |<-------------------------- | | ****** ( missing part) | 200 ok | | |--------------------------> | | As on the 103, auth windows pop up, then click ok to accept (thus it will added 101 as contact), in this case it does not do any xcap operation? Is it the correct call flow? Thanks min On 06/19/2012 04:49 PM, Min Wang wrote: > Hi > > Here is another similar case which does not work. maybe it is related > the above issue as well. > > Jitsi 1.1 client: 101/103 , with xcap as storage > proxy: kamailio 3.3 > ip network is: 192.168.122.0 > > the setup is like the following: > > 101(.224) --- proxy (.32) --- 103 (.1) > > After all presence related db are truncated, kamailio was restart. > then bring 101,103 online. > > On 101 add 103, 103 prompt up authorization window, click ok. > now on 103, I can see 101 online status > > but on 101, I can not see 103 status <----the first trace file: the > jitsi-101-add-103-confirm-2.pcap > > (it is grey, even manually change 103 status from online to away > etc, on 101, the 103 always showed grey. <--- second trace file: > jitsi-101-103-change.pcap) > > Please see the attached pcap files. > > > the watcher table shows: > > id presentity_uri watcher_username watcher_domain > event status reason inserted_time > 1 sip:103@192.168.122.32 101 192.168.122.32 > presence 2 1340114579 > 2 sip:101@192.168.122.32 103 192.168.122.32 > presence 1 1340114582 > > the active wacher table show: > > id presentity_uri watcher_username watcher_domain > to_user to_domain event event_id to_tag from_tag > callid local_cseq remote_cseq contact record_route > expires status reason version socket_info > local_contact from_user from_domain updated updated_winfo > 1 sip:101@192.168.122.32 101 192.168.122.32 > 101 192.168.122.32 message-summary > a6a1c5f60faecf035a1ae5b6e96e979a-7e2e 8b6ca8e7 > 9e8288d04d0dab18fda28f5fb24b40d8(a)0.0.0.0 1 2 > sip:101@192.168.122.224:5060;transport=udp;registe... > 1340117922 1 1 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 101 192.168.122.32 -1 -1 > 2 sip:101@192.168.122.32 101 192.168.122.32 > 101 192.168.122.32 presence.winfo > a6a1c5f60faecf035a1ae5b6e96e979a-8a99 9975004d > 9b8cb0735bc559232639aa28c16a6f4b(a)0.0.0.0 2 2 > sip:101@192.168.122.224:5060;transport=udp;registe... > 1340117921 1 2 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 101 192.168.122.32 -1 -1 > 3 sip:103@192.168.122.32 103 192.168.122.32 > 103 192.168.122.32 message-summary > a6a1c5f60faecf035a1ae5b6e96e979a-0117 9d873afb > 30fa1ce8cca7a7c50fea11e688ec3d5e@0:0:0:0:0:0:0:0 1 2 > sip:103@192.168.122.1:5060;transport=udp;registeri... > 1340117959 1 1 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 103 192.168.122.32 -1 -1 > 4 sip:103@192.168.122.32 103 192.168.122.32 > 103 192.168.122.32 presence.winfo > a6a1c5f60faecf035a1ae5b6e96e979a-9a71 101d721c > 148be0ec1a4acc087c0083324398cf14@0:0:0:0:0:0:0:0 2 2 > sip:103@192.168.122.1:5060;transport=udp;registeri... > 1340117959 1 2 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 103 192.168.122.32 -1 -1 > 5 sip:103@192.168.122.32 101 192.168.122.32 > 103 192.168.122.32 presence > a6a1c5f60faecf035a1ae5b6e96e979a-5ab7 e9099546 > 6ed81a0fa2042996295974b671a8074f(a)0.0.0.0 1 2 > sip:101@192.168.122.224:5060;transport=udp;registe... > 1340118179 2 1 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 101 192.168.122.32 -1 -1 > 6 sip:101@192.168.122.32 103 192.168.122.32 > 101 192.168.122.32 presence > a6a1c5f60faecf035a1ae5b6e96e979a-98a8 > > > > > thanks > > min > > On 06/19/2012 01:00 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> you have to share the content of the messages sent, providing just >> the url is not enough to see what happens. >> >> watchers table is for the list of contacts and their global status in >> regard to presentity, active_watchers is for keeping the presence >> dialogs. >> >> Cheers, >> Daniel >> >> On 6/19/12 12:47 PM, Min Wang wrote: >>> Hi >>> >>> (1) I tested more. It is due to the difference between bria 3 and >>> jitsi while handling the adding/removing contact: >>> >>> if contact is added/removed: >>> >>> for jitsi, it send out: >>> >>> PUT /xcap-root/resource-lists/users/sip:101@192.168.122.32/index >>> PUT /xcap-root/pres-rules/users/sip:101@192.168.122.32/presrules >>> >>> >>> for bria 3, it only send out >>> >>> PUT >>> /xcap-root/resource-lists/users/101(a)192.168.122.32/contacts-resource-list.xml >>> HTTP/1.0. >>> PUT >>> /xcap-root/resource-lists/users/101(a)192.168.122.32/contacts-resource-list.xml >>> >>> >>> it does NOT send out the pres-rules. >>> >>> Is it the good behavior ( from RFC point of view) ? >>> should I call the pres_update_watchers even though its is for >>> resource-lists? >>> >>> (2) BTW, what is the exact purpose of watcher table? >>> >>> e.g: on 101, if delete 103, watcher table become: >>> >>> id presentity_uri watcher_username >>> watcher_domain event status reason inserted_time >>> 1 sip:103@192.168.122.32 101 192.168.122.32 >>> presence 1 1340096051 >>> 2 sip:101@192.168.122.32 103 192.168.122.32 >>> presence 3 terminated 1340096181 >>> >>> it is item 2 become terminated instead of item 1. If watcher >>> table is for subscription, then seems item 1 should be terminated? >>> Look at the code, it seems somehow serve as mixed role of >>> subscription/auth/xcap purpose, not for subscription state? >>> >>> and is the active_watcher mainly for the state machine of >>> http://tools.ietf.org/html/rfc3857#section-4.7.1? >>> >>> thanks. >>> >>> >>> min >>> >>> >>> >>> >>> >>> >>> >>> >>> On 06/19/2012 10:41 AM, Daniel-Constantin Mierla wrote: >>>> Hello, >>>> >>>> On 6/18/12 2:11 PM, Andreas Granig wrote: >>>>> Hi, >>>>> >>>>> On 06/15/2012 05:25 PM, Min Wang wrote: >>>>>> | 1 | sip:103@192.168.122.32 | 101 | 192.168.122.32 | >>>>>> presence | 2 | | 1339772803 | >>>>>> >>>>>> then I deleted the 103 from the contact list, the watcher >>>>>> table still >>>>>> shows the same. >>>>> For local storage, I'd expect an unsubscribe (subscribe with >>>>> Expires=0) >>>>> from 101 to 103. The obvious work-flow would be to set it to >>>>> "terminated" in watchers table, and in case of a subsequent >>>>> re-subscribe >>>>> it should be changed back to "pending" state, although the >>>>> state-machine >>>>> doesn't indicate a transition from "terminated" back to "pending". >>>>> Isn't >>>>> this the case? >>>>> >>>>> For xcap storage, there are other ways to block/remove a contact >>>>> on/from >>>>> the list. As Iñaki pointed out in >>>>> http://lists.opensips.org/pipermail/devel/2009-August/003868.html, >>>>> the >>>>> xcap server needs to notify the sip server about the change, which in >>>>> turn will notify the other party (103) that it's no longer allowed to >>>>> see 101's state. If the xcap_server module of kamailio is used, >>>>> there is >>>>> the following code snippet in some examples floating around on the >>>>> internet: >>>>> >>>>> >>>>> switch($rm) { >>>>> case "PUT": >>>>> xcaps_put("$var(uri)", "$hu", "$rb"); >>>>> if($xcapuri(u=>auid)=~"pres-rules") { >>>>> pres_update_watchers("$var(uri)", "presence"); >>>>> pres_refresh_watchers("$var(uri)", "presence", 1); >>>>> } >>>>> >>>>> So, shouldn't this update the watchers accordingly? Anyways, also in >>>>> this case the watcher state should change to "terminated", and in >>>>> case >>>>> of a re-subscribe it should go back to pending if xcap rules are >>>>> allowing this. >>>>> >>>>> Maybe someone with good xcap_server/presence insights could >>>>> elaborate on >>>>> that? >>>> to summarize, you say that when the contact is removed from the >>>> buddy list, the watcher table is not updated to state terminated by >>>> pres_update_watchers(...)? >>>> >>>> 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 0

About the subscribe/200-ok contact address on kamailio 3.3
by Min Wang 19 Jun '12

19 Jun '12

HI I have tried the kamailio 3.3, user 101(10.15.20.131) send subscribe to (10.15.20.137), kamailio return 200 OK, but with contact header as: Contact: <sip:10.15.20.137:5060;transport=udp>. is it correct? Since before subscription expired, the client will send the re-subscribe using that contact address, is it right? Should kamailio return the contact as sip:101@10.15.20.137 in this case? Thanks. min ///////////////////////////// pkt captured //// U 10.15.20.131:54740 -> 10.15.20.137:5060 SUBSCRIBE sip:101@10.15.20.137 SIP/2.0. Via: SIP/2.0/UDP 192.168.122.224:54740;branch=z9hG4bK-d8754z-87f9c58f1c70cbe2-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:101@10.15.20.131:54740>. To: <sip:101@10.15.20.137>. From: "101"<sip:101@10.15.20.137>;tag=4f7671df. Call-ID: MWRjNjU2NGNmMWE0Nzk3ODU2ZTZjZWViYmViYmQ1NmY. CSeq: 2 SUBSCRIBE. Expires: 60. Accept: application/watcherinfo+xml. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Proxy-Authorization: Digest username="101",realm="10.15.20.137",nonce="T9n8u0/Z+4/OOU2Qeo4acn7OE+7pb5A1",uri="sip:101@10.15.20.137",response="93caab1f09daa88a1619bb0613b9a5ff",algorithm=MD5. Supported: eventlist. User-Agent: Bria 3 release 3.4.1 stamp 66953. Event: presence.winfo. Content-Length: 0. . # U 10.15.20.137:5060 -> 10.15.20.131:54740 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.122.224:54740;branch=z9hG4bK-d8754z-87f9c58f1c70cbe2-1---d8754z-;rport=54740;received=10.15.20.131. To: <sip:101@10.15.20.137>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-21e3. From: "101"<sip:101@10.15.20.137>;tag=4f7671df. Call-ID: MWRjNjU2NGNmMWE0Nzk3ODU2ZTZjZWViYmViYmQ1NmY. CSeq: 2 SUBSCRIBE. Expires: 60. Contact: <sip:10.15.20.137:5060;transport=udp>. Server: kamailio (3.3.0-rc0 (x86_64/linux)). Content-Length: 0. . .... U 10.15.20.131:54740 -> 10.15.20.137:5060 SUBSCRIBE sip:10.15.20.137:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.122.224:54740;branch=z9hG4bK-d8754z-59fd0765744f37a3-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:101@10.15.20.131:54740>. To: <sip:101@10.15.20.137>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-21e3. From: "101"<sip:101@10.15.20.137>;tag=4f7671df. Call-ID: MWRjNjU2NGNmMWE0Nzk3ODU2ZTZjZWViYmViYmQ1NmY. CSeq: 8 SUBSCRIBE. Expires: 60. Accept: application/watcherinfo+xml. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Proxy-Authorization: Digest username="101",realm="10.15.20.137",nonce="T9n8u0/Z+4/OOU2Qeo4acn7OE+7pb5A1",uri="sip:10.15.20.137:5060;transport=udp",response="f9ef9d95891bbe5722bb47fbe93b20e9",algorithm=MD5. Supported: eventlist. User-Agent: Bria 3 release 3.4.1 stamp 66953. Event: presence.winfo. Content-Length: 0.

2 1

IP ToS
by Olle E. Johansson 19 Jun '12

19 Jun '12

I know that you can set the TOS for packets sent from Kamailio in the config section. I got the question if you can read the TOS in the IP packet of an incoming response or reply in the config scripts? And can you set it per message/transaction? /O

2 1

install problem
by copycall 19 Jun '12

19 Jun '12

hello, i ran into a problem installing kamailio when i got to #7 "create mysql database": < openser.cfg.5 > /usr/local/share/man//man5/ openser.cfg.5 chmod 644 /usr/local/share/man//man5/openser.cfg.5 sed -e "s#/etc/openser/openser\.cfg#/usr/local/etc/openser/openser.cfg#g" \ -e "s#/usr/sbin/#/usr/local/sbin/#g" \ -e "s#/usr/lib/openser/modules/#/ usr/local/lib64/openser/modules/#g" \ -e "s#/usr/share/doc/openser/#/ usr/local/share/doc/openser/#g" \ < scripts/openserctl.8 > /usr/local/share/man//man8/ openserctl.8 chmod 644 /usr/local/share/man//man8/openserctl.8 sed -e "s#/etc/openser/openser\.cfg#/usr/local/etc/openser/openser.cfg#g" \ -e "s#/usr/sbin/#/usr/local/sbin/#g" \ -e "s#/usr/lib/openser/modules/#/ usr/local/lib64/openser/modules/#g" \ -e "s#/usr/share/doc/openser/#/ usr/local/share/doc/openser/#g" \ < utils/openserunix/openserunix.8 > \ /usr/local/share/man//man8/openserunix.8 chmod 644 /usr/local/share/man//man8/openserunix.8 root@copycall:/usr/local/src/openser-1.3.0/sip-server# echo $PATH /usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin root@copycall:/usr/local/src/openser-1.3.0/sip-server# cd root@copycall:~# /usr/local/etc/kamailio/kamctlrc bash: /usr/local/etc/kamailio/kamctlrc: No such file or directory root@copycall:~# /usr/local/sbin/kamdbctl create bash: /usr/local/sbin/kamdbctl: No such file or directory any solution will be greatly appreciated, please include all the code as kamailio is new to me. thank you in advance, dave

2 1

Releasing v3.3.0 later today
by Daniel-Constantin Mierla 18 Jun '12

18 Jun '12

Hello, quick note about packaging the 3.3.0 today, therefore if anyone has commits to push to GIT branch 3.3 after 12:00GMT, please write first to sr-dev mailing list. Once the announcement about the release is sent out, the commits can be pushed again directly. 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

2 1

Sipcapture and Siptrace Compile Error
by Akan 18 Jun '12

18 Jun '12

When compiling kamailio v3.3, I am getting this error from sipcapture and siptrace. I am compiling on solaris 10, 64bit Sun server with gnu tools. CC (gcc) [M sipcapture.so] sipcapture.o In file included from sipcapture.c:76:0: sipcapture.h:48:4: error: unknown type name 'u_int32_t' sipcapture.h:49:4: error: unknown type name 'u_int32_t' gmake[1]: *** [sipcapture.o] Error 1 In file included from siptrace.c:54:0: ../../modules/sipcapture/sipcapture.h:48:4: error: unknown type name 'u_int32_t' ../../modules/sipcapture/sipcapture.h:49:4: error: unknown type name 'u_int32_t' gmake[1]: *** [siptrace.o] Error 1 Thanks Nathaniel

2 4

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Charles Chance 18 Jun '12

18 Jun '12

Hello Daniel, Sorry for the delay! I will get around to testing with the latest master branch sometime over the next couple of days and will report back here. Many thanks, Charles Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Hello, can you try with latest master branch -- it should be fixed by: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=1d89d7b… Cheers, Daniel On 5/29/12 5:52 PM, Daniel-Constantin Mierla wrote: Hello, finally had a chance to look over it. I guess it is some race, resulting in a memory space being overwritten. The 200ok for publish with ringing state is processed more or less at the time when 200ok for invite comes and dialog changes state. Can you re-compile with MEMDBG=1 (edit the Makefile.defs or give it in the command line parameter) and try again? Maybe the memory manager can catch the fault operation and provide more details. Cheers, Daniel On 5/17/12 6:58 PM, Charles Chance wrote: Hi Daniel, I have sent the messages over to you. I look forward to hear what you can find. Best, Charles From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: 17 May 2012 15:06 To: Charles Chance Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List' Subject: Re: FW: (Devel) Segmentation fault using pua_dialoginfo Hello, the content of ps->param is pretty much all invalid. So looks like a callback on a freed/invalid parameter, could be a double-callback execution. Set debug=3 and send all the messages for such case -- it might be quite a lot of them, so send them to my email address, the mailing list has a limit of message size. Cheers, Daniel On 5/17/12 2:13 PM, Charles Chance wrote: Hi, Requested output as follows: (gdb) frame 1 #1 publ_cback_func (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) at send_publish.c:246 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); (gdb) p *ps $2 = {req = 0x0, rpl = 0xb7c30340, param = 0xb3dc3ff4, code = 412, flags = 0, branch = 0, t_rbuf = 0x0, dst = 0x0, send_buf = {s = 0x0, len = 0}} (gdb) p *(ua_pres_t*)(*ps->param) $3 = {id = {s = 0x20455942 <Address 0x20455942 out of bounds>, len = 980445555}, pres_uri = 0x40323531, event = 875444279, expires = 875703856, desired_expires = 858861105, flag = 808794676, db_flag = 1394618421, cb_param = 0x322f5049, next = 0xa0d302e, ua_flag = 979462486, etag = { s = 0x50495320 <Address 0x50495320 out of bounds>, len = 808333871}, tuple_id = {s = 0x5044552f <Address 0x5044552f out of bounds>, len = 775436064}, body = 0x322e3034, content_type = { s = 0x312e3134 <Address 0x312e3134 out of bounds>, len = 1648047155}, watcher_uri = 0x636e6172, call_id = { s = 0x397a3d68 <Address 0x397a3d68 out of bounds>, len = 1647593320}, to_tag = {s = 0x3435634b <Address 0x3435634b out of bounds>, len = 842149473}, from_tag = { s = 0x63306461 <Address 0x63306461 out of bounds>, len = 808334648}, cseq = 1767246349, version = 1394621025, outbound_proxy = 0x322f5049, extra_headers = 0x552f302e, record_route = { s = 0x31205044 <Address 0x31205044 out of bounds>, len = 825111097}, remote_contact = {s = 0x312e3836 <Address 0x312e3836 out of bounds>, len = 976498224}, contact = { s = 0x36373135 <Address 0x36373135 out of bounds>, len = 1701985073}} Yes, it is a test server so very happy to arrange for remote access if required. In the meantime, I will do a little more digging to try to find out why entity is null. Cheers, Charles -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2176 / Virus Database: 2425/5004 - Release Date: 05/16/12 -- 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 -- 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 0

Conditionnal hunting with usrloc?
by Igor Potjevlesch 15 Jun '12

15 Jun '12

Hi all, I'm facing a problem with multiple contact in 'location' table with the same 'username'. The idea is to rewrite R-URI with 1 contact and if there is no answer or SIP 5xx, try with the second. Is that make sense? I use SER0.9.9. Regards, Igor.

3 5

Architecture
by Uriel Rozenbaum 15 Jun '12

15 Jun '12

Hey guys, I'm trying to plan a deployment for a couple thousands of users. As we are facing now some issues for scaling, I'd like to make a modular architecture in order to add a server when needed. I was thinking about N+1 Kamailio Proxies to Register customers and handle the border of the network, 1+1 Redirect servers so internal Network Elements can "find" the right Proxy to terminate a call. Customers would use DNS SRV to decide which Proxy to send a REGISTER and then send INVITES to that same proxy. As we have a private core network addressing, I'm thinking I'll have an Asterisk farm in order to perform media handling. Do you have some recommendation or guidelines for this deployment? Thanks! Uriel

3 3

AMR and AMR-WB support
by Jakson Kalsson 15 Jun '12

15 Jun '12

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the SIPX support these codecs and RFC4867 ? Also, any other Server/PBX which support AMR, AMR-WB recommended are welcome. Best regards,

3 3

add user/delete/add user again on kamailio 3.3 presence server
by Min Wang 15 Jun '12

15 Jun '12

HI I am testing the following case in the kamailio 3.3 presence server, all are in 192.168.122.0 network 101 (bria, ip:.224) ----- kamailio ps( ip:.32) --- 103 ( jitsi, ip:.1) on 101 I added 103 in the contact list, 101 <-> 103 can see each other, Then on 101, I deleted 103, thus on 103, 101 became grey, so far so good. Now I tried to add 103 back to contact list, on 101 I can see 103's status (online), but on jitsi 103, I can not see 101's status. ( grey) The watcher table showed that status terminated. +----+------------------------+------------------+----------------+----------+--------+------------+---------------+ | id | presentity_uri | watcher_username | watcher_domain | event | status | reason | inserted_time | +----+------------------------+------------------+----------------+----------+--------+------------+---------------+ | 1 | sip:103@192.168.122.32 | 101 | 192.168.122.32 | presence | 1 | | 1339752028 | | 2 | sip:101@192.168.122.32 | 103 | 192.168.122.32 | presence | 3 | terminated | 1339752031 | +----+------------------------+------------------+----------------+----------+--------+------------+---------------+ I guess it should be status=1 in this case, but somehow kamailio seems does not update the table? I have a complete packet-trace file if needed. thanks. min

1 0

drouting module: can I use non-digit prefix?
by Yufei Tao 14 Jun '12

14 Jun '12

Hi I'm using drouting module and seems that it only allows digit-only prefixes. Also all the number of digits to be stripped ('strip' defined in dr_gateways) must be all digits too. When I used non-digits in these, Kamailio crashed. Is there any way in drouting module that I can use prefixes with alphabet and dots, e.g. "gateway1."? Thanks very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

2 7

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Daniel-Constantin Mierla 14 Jun '12

14 Jun '12

Hello, finally had a chance to look over it. I guess it is some race, resulting in a memory space being overwritten. The 200ok for publish with ringing state is processed more or less at the time when 200ok for invite comes and dialog changes state. Can you re-compile with MEMDBG=1 (edit the Makefile.defs or give it in the command line parameter) and try again? Maybe the memory manager can catch the fault operation and provide more details. Cheers, Daniel On 5/17/12 6:58 PM, Charles Chance wrote: > > Hi Daniel, > > I have sent the messages over to you. > > I look forward to hear what you can find. > > Best, > > Charles > > ------------------------------------------------------------------------ > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* 17 May 2012 15:06 > *To:* Charles Chance > *Cc:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) > -Users Mailing List' > *Subject:* Re: FW: (Devel) Segmentation fault using pua_dialoginfo > > Hello, > > the content of ps->param is pretty much all invalid. So looks like a > callback on a freed/invalid parameter, could be a double-callback > execution. > > Set debug=3 and send all the messages for such case -- it might be > quite a lot of them, so send them to my email address, the mailing > list has a limit of message size. > > Cheers, > Daniel > > On 5/17/12 2:13 PM, Charles Chance wrote: > >> Hi, >> >> Requested output as follows: >> >> (gdb) frame 1 >> >> #1 publ_cback_func (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) >> >> at send_publish.c:246 >> >> 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); >> >> (gdb) p *ps >> >> $2 = {req = 0x0, rpl = 0xb7c30340, param = 0xb3dc3ff4, code = 412, >> flags = 0, >> >> branch = 0, t_rbuf = 0x0, dst = 0x0, send_buf = {s = 0x0, len = 0}} >> >> (gdb) p *(ua_pres_t*)(*ps->param) >> >> $3 = {id = {s = 0x20455942 <Address 0x20455942 out of bounds>, >> >> len = 980445555}, pres_uri = 0x40323531, event = 875444279, >> >> expires = 875703856, desired_expires = 858861105, flag = 808794676, >> >> db_flag = 1394618421, cb_param = 0x322f5049, next = 0xa0d302e, >> >> ua_flag = 979462486, etag = { >> >> s = 0x50495320 <Address 0x50495320 out of bounds>, len = 808333871}, >> >> tuple_id = {s = 0x5044552f <Address 0x5044552f out of bounds>, >> >> len = 775436064}, body = 0x322e3034, content_type = { >> >> s = 0x312e3134 <Address 0x312e3134 out of bounds>, len = 1648047155}, >> >> watcher_uri = 0x636e6172, call_id = { >> >> s = 0x397a3d68 <Address 0x397a3d68 out of bounds>, len = 1647593320}, >> >> to_tag = {s = 0x3435634b <Address 0x3435634b out of bounds>, >> >> len = 842149473}, from_tag = { >> >> s = 0x63306461 <Address 0x63306461 out of bounds>, len = 808334648}, >> >> cseq = 1767246349, version = 1394621025, outbound_proxy = 0x322f5049, >> >> extra_headers = 0x552f302e, record_route = { >> >> s = 0x31205044 <Address 0x31205044 out of bounds>, len = 825111097}, >> >> remote_contact = {s = 0x312e3836 <Address 0x312e3836 out of bounds>, >> >> len = 976498224}, contact = { >> >> s = 0x36373135 <Address 0x36373135 out of bounds>, len = 1701985073}} >> >> Yes, it is a test server so very happy to arrange for remote access >> if required. In the meantime, I will do a little more digging to try >> to find out why entity is null. >> >> Cheers, >> >> Charles >> > > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > > ------------------------------------------------------------------------ > > No virus found in this message. > Checked by AVG - www.avg.com <http://www.avg.com> > Version: 2012.0.2176 / Virus Database: 2425/5004 - Release Date: 05/16/12 > -- 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

1 1

Making dialog module work for me !!
by SamyGo 14 Jun '12

14 Jun '12

Hello I've two problems regarding dialog module- used for profiling users that are in-a-call. My topology is quiet simple. SIP-USER-71141---->Kamailio------------->Asterisk[AGI]|| CALLEE<-----------------Kamailio<--------------------------------|| 1- I wish to store my custom SIP header value in profiles. I'm having trouble in figuring out on how to use Dialog_vars table in my kamailio server. I'm using dialog module to profile SIP users that are in call or making a call. It works well and the dialog maintains an entry when the call is established. Now, the difficult part is that I want my custom header to be inserted into either dialog table or dialog_vars table such that when I do a query "*is_in_profile("$avp(my_value)")*" it returns me a result. 2- Alternative to 1, I can log A,B leg separately but it doesn't erase all data on call hangup. Asterisk AGI changes the caller-id of SIP-USERA such that if it was 71141 it would be something like "007" on its way out to destination via kamailio. Now, at this point in kamailio when the call is answered by destination the dialog module saves the call data with values corresponding to "007" and not "71141". I have to disable the "dialog", "detect_spirals" in order to count this one call as two separate entities BUT problem occurs when on hangup the B-Leg gets erased from the table but the A-leg remains in DB for as long as infinity. Please help me in this. Let me know if more details are required from me. Regards, Sammy G.

2 9

Kamailio debian repository changes
by Jon Bonilla 14 Jun '12

14 Jun '12

Hi All With the upcoming release of version 3.3.0, we have updated our kamailio debian repositories. * Debian 5.0 "lenny" will no longer be supported. This means that no more nightly builds for this distribution will be triggered. We'll provide builds for stable releases of 3.0 ,3.1 and 3.2 branches though. * Ubuntu 12.04 "Precise" is now supported for nightly builds and 3.2+ versions. Next time a 3.2.x version is tagged we'll provide the build. Meanwhile you can download the nightly build of 3.2 branch for this distro. * There will be no more nightly builds of 3.1 branch. We'll provide stable builds in case another version is released though. * We now support nightly builds of branch 3.3 (same as 3.2 and master). You can test latest git versions of branch 3.3 until a stable release is built using these repositories: deb http://deb.kamailio.org/kamailio33-nightly squeeze main deb http://deb.kamailio.org/kamailio33-nightly wheezy main deb http://deb.kamailio.org/kamailio33-nightly lucid main deb http://deb.kamailio.org/kamailio33-nightly precise main All the information and repositories can be found in kamailio wiki as usual: http://www.kamailio.org/dokuwiki/doku.php/packages:debs http://www.kamailio.org/wiki/packages/debs

1 0

Dispatcher alg:7 not working accordingly
by SamyGo 14 Jun '12

14 Jun '12

Hello, I'm having trouble using algorithm 7 in dispatcher module. Here is my kamailio version. The problem is that despite multiple dispatcher entries corresponding to one set all calls go only to only one destination. [root@control1 ~]# kamailio -V version: kamailio 3.2.3 (x86_64/linux) 59f87e 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: 59f87e compiled on 18:04:08 Apr 19 2012 with gcc 4.1.2 [root@control1 ~]# Here is the dipacther module params. # ------- Load-balancer params ------ modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@109.XXX.2XX.XX") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") modparam("dispatcher", "hash_pvar", "$avp(i:274)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "ds_hash_initexpire", 60) Below is the function call based on avp(i:274) avp_db_query("select uid from uid_did_map where accountcode = '$fU'","$avp(i:274)"); xlog("L_INFO","$rm from $fu (IP:$si:$sp) --------- UID:'$avp(i:274)-----------'\n"); ds_load_update(); if(!ds_select_dst("1","7")) { sl_send_reply("500", "No destination available"); xlog("route[MAIN] : $rm : No destinations available for $rd"); exit; } I'm getting two different values for avp(i:274) {26,29} but the destination are not hashed on its base. Please help. Regards, Sammy G.

3 15

Alias possible for presence...
by Gertjan Wolzak 14 Jun '12

14 Jun '12

Hello All, I have the following challenge. We are using kamailio 3.2 and have enabled presence. Which works fine. But we would like to be able to work with aliases, so that the account name and "presentation" name can be different. Any Ideas. Or did I miss the documentation on that? Regards, Gertjan Wolzak

1 0

Lua script calling rewriteuri
by Sharif Tanvir Rahman 14 Jun '12

14 Jun '12

Hello, We plan to use Kamailio as redirect server. Corresponding contact list is to be fetched by a Lua script from DB. The problem is about calling the required 'rewriteuri' function: 1. From lua script. both 'sr.rewriteuri()' and 'sr.modf("rewriteuri", ...)" fails to write the 'contact' field. 2. I tried an alternative way; pushing the value from lua (e.g. sr.pv.sets("contact", "sip:...")) and got the value in the Kamailio.cfg ($var(contact)). But can't call the rewriteuri function with this variable's value as argument. This seems to be a general problem of calling a function from .cfg with a variable's value as parameter. Any help much appreciated. -- Sharif

2 3

topoh - capturing encrypted values
by Alex Balashov 13 Jun '12

13 Jun '12

Given the transparency with which 'topoh' operates, is there any way to capture the new, fake values that it has substituted? In other words, can I, for example, xlog the new Call-ID produced by it? Or does that all take place on a level of abstraction that cannot be intercepted in script? Thanks, -- 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/

2 1

About xcap xuid in kamailio
by Min Wang 13 Jun '12

13 Jun '12

Hi I am trying to use kamailio xcap server, saw the sample code: if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; why should the xuid have sip:xx@domain.com format? The xcap standard seems does not enforce the format on the xui part? http://xcap.example.com/root/resource-lists/users/joe |-------------v-------------||------v------||---v---| document root AUID XUI thanks. min

4 14

AVPs - how many values on stack
by Olle E. Johansson 13 Jun '12

13 Jun '12

I got this question today that I can't find the answer on: How many values are there in an avp? If I have pushed X values into the AVP - is there a function that counts them for me? /O

2 1

release of v3.3.0 scheduled for Monday, June 18, 2012
by Daniel-Constantin Mierla 13 Jun '12

13 Jun '12

Hello, as branch 3.3 was created and several web pages updated/created already for v3.3.0, the plan is to make the full release next Monday, on June 18, 2012. Packaging specs look very good thanks to Jon and Peter. The few issues left open should be worked meanwhile. For other opinions, reply with your comments here. 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 0

About sdpops
by Grégoire Vandendeurpel 13 Jun '12

13 Jun '12

Hello, I'm using Kamailio 3.2.3 and SDPOPS module. I just have a question about the configuration: What do you mean by "ANY_ROUTE" in configuration ?? Cause I don't know where in the configuration , I have to use the differents functions of SDPOPS. If I do something like this: request_route { if ($fU == "103") { sdp_remove_codecs_by_id("0,3,8"); #sdp_remove_media("audio"); } It does not work . So please help me about where I have to put this in?? Thanks in advance -- *Grégoire Vandendeurpel, * * * *IT Sector*

2 8

sipcapture logging
by Konstantin M. 13 Jun '12

13 Jun '12

Hello, Is it possible to log SIP sessions using sipcapture module to log files by log format instead of database ? Thanks!

3 6

Re: [SR-Users] Kamailio Unexpectedly Terminating
by Nathaniel L. Keeling III 12 Jun '12

12 Jun '12

Yes, This is a 64bit system T-Mobile. America’s First Nationwide 4G Network Jason Penton <jason.penton(a)smilecoms.com> wrote: >from the logs he has sent the mem addresses all look 64 bit. Sparc AFAIK is >only 64 bit anyway > >Cheers >Jason > >On Tue, Jun 12, 2012 at 8:50 AM, Daniel-Constantin Mierla <miconda(a)gmail.com >> wrote: > >> Hello, >> >> is it 64bit architecture? >> >> Cheers, >> Daniel >> >> >> On 6/11/12 9:12 PM, Akan wrote: >> >>> Here is the information requested: >>> >>> System = SunOS >>> Node = -f >>> Release = 5.10 >>> KernelID = Generic_141444-09 >>> Machine = sun4u >>> BusType = <unknown> >>> Serial = <unknown> >>> Users = <unknown> >>> OEM# = 0 >>> Origin# = 1 >>> NumCPU = 1 >>> >>> One machine is a Sun Fire V120, 2g memory, UltraSPARC-IIe 650MHz, >>> UltraAX-i2 >>> The other machine is a Sun Netra T1 200, 1g memory, UltraSparc-IIe >>> 500MHz, UltraAX-i2 >>> >>> Thanks >>> >>> Nathaniel L Keeling >>> >>> >>> >>> On 6/11/2012 8:33 AM, Daniel-Constantin Mierla wrote: >>> >>>> Hello, >>>> >>>> I committed a patch that should make it work when realm_prefix is not >>>> set for registrar module: >>>> >>>> http://git.sip-router.org/cgi-**bin/gitweb.cgi/sip-router/?a=**commit;h= >>>> **d42379da90f2ec87cb5dbb00ebb563**c7528ec910<http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=d42379d…> >>>> >>>> For the moment I have no idea why SIGBUS is triggered, i tried to access >>>> the structure directly, not via defines, and there is still a problem when >>>> realm_prefix is set. It's ok accessing the .len field but not .s of the >>>> same structure. The field .s is the first, making me think it's not >>>> alignment issue, or it is something very specific to this system. >>>> >>>> Overall, I am not familiar with sparc/solaris systems, maybe if you send >>>> here full details about OS version, architecture and CPU type, others using >>>> similar systems can give hints. >>>> >>>> Jason reported that on solaris with x86, this problem was not >>>> encountered. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On 6/4/12 8:57 AM, Daniel-Constantin Mierla wrote: >>>> >>>>> Hello, >>>>> >>>>> still strange with the crash on that if condition. One of your previous >>>>> reports showed a different line, when assigning a pointer. Also, I kind of >>>>> understood that save() was ok, but lookup() not. Is not the case anymore as >>>>> the backtrace shows. >>>>> >>>>> SIGBUS can occur because of alignment in memory which is thrown in >>>>> solaris, does not happen in linux flavours. I am thinking of putting the >>>>> prefix field first so the access to it is at beginning of the straucture, >>>>> just to see if that is the problem, although is should not be. >>>>> >>>>> I will add a patch soon, otherwise I am out of ideas for the moment >>>>> with the SIGBUS, I would need access to the machine to do some testing by >>>>> myself. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> On 6/2/12 10:36 PM, Akan wrote: >>>>> >>>>>> Ok, I reloaded the servers with v3.2 from git without performing the >>>>>> checkout and with just the master branch. Reran my tests on 2 servers and >>>>>> Kamailio terminated with a core dump. I have included the full trace of one >>>>>> of the servers. The other trace has the same results. >>>>>> >>>>>> Thanks >>>>>> >>>>>> Nathaniel >>>>>> >>>>>> On 5/30/2012 2:25 PM, Daniel-Constantin Mierla wrote: >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> On 5/30/12 9:10 PM, Akan wrote: >>>>>>> >>>>>>>> This was a re-pull from the git master. I had deleted all of the old >>>>>>>> files and issued these commands: >>>>>>>> >>>>>>>> git clone --depth 1 git://git.sip-router.org/sip-**router<http://git.sip-router.org/sip-router>kamailio >>>>>>>> cd kamailio >>>>>>>> git checkout -b 3.2 origin/3.2 >>>>>>>> >>>>>>>> This is the source that I got. Could this be a problem within git? >>>>>>>> >>>>>>> do not checkout branch 3.2, just try with the master branch, as said, >>>>>>> not all patches are backported to stable 3.2. >>>>>>> >>>>>>> For sake of completeness, here are some guidelines: >>>>>>> http://www.kamailio.org/wiki/**install/devel/git<http://www.kamailio.org/wiki/install/devel/git> >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> >>>>>> >>>>>> ______________________________**_________________ >>>>>> 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<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>>>>> >>>>> >>>>> -- >>>>> Daniel-Constantin Mierla -http://www.asipto.com >>>>> http://twitter.com/#!/miconda -http://www.linkedin.com/in/**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 >>>>> >>>>> >>>>> >>>> -- >>>> Daniel-Constantin Mierla -http://www.asipto.com >>>> http://twitter.com/#!/miconda -http://www.linkedin.com/in/**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 >>>> >>>> >>>> No virus found in this message. >>>> Checked by AVG - www.avg.com <http://www.avg.com> >>>> Version: 2012.0.2178 / Virus Database: 2433/5061 - Release Date: 06/10/12 >>>> >>>> >>> >>> >>> ______________________________**_________________ >>> 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<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.com >> http://twitter.com/#!/miconda - http://www.linkedin.com/in/**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 >> >> >> >> >> ______________________________**_________________ >> 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<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >> > >This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer > >_______________________________________________ >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

High availability design with Kamailio
by Reinhard, Carel (GD) 12 Jun '12

12 Jun '12

Dear all, hope to get some information from you guys! For a redundant service we have to build up SIP proxy/registrar server in a high available scenario. Therefore, we will have 2 SIP proxy/registrar on two completely independent servers. They should be SIP dialog stateful and replicate all the session/call states between each other. If one server fails, the other should have all the SIP dialog information for call handling and accounting. How can this done in a stable and reliable way? Is this feature mature enough to support enterprise requirements? The only information I have is to build it up with the DISPATCHER module, but it seems more to be a stateless load-balancer instead of a HA module. Who has experience with HA failover designs with Kamailio and can give me some hints? Thank you very much in advance Kind regards Carel Carel Reinhard Security and VoIP Engineer SRG SSR Switzerland Telefon direkt +41 31 350 94 07 Carel.Reinhard(a)srgssr.ch<mailto:Carel.Reinhard@srgssr.ch>

5 10

Shared memory fragmentation
by Morten Isaksen 12 Jun '12

12 Jun '12

Hi, We have 2 Kamailio 3.0.3 servers that has been running with carrierroute for about 2 years without any problems. They have 128 MB shared memory and modparam("carrierroute", "config_source", "db") modparam("carrierroute", "db_url", "<DBURL>") modparam("carrierroute", "fetch_rows", 500) The carrierroute table is about 91K lines and have been growing slowly. Suddenly we get this ERROR: carrierroute [cr_data.c:585]: could not allocate shared memory from available pool after a few "kamctl cr reload". I increased the shared memory to 256 MB but with the same result. I have now increased it to 512 MB and it seems to work better now. I have noticed this. After a restart the shmem counters is like this: shmem:total_size = 536870912 shmem:used_size = 28486752 shmem:real_used_size = 40147128 shmem:max_used_size = 41135424 shmem:free_size = 496723784 shmem:fragments = 555 And after the first "kamctl cr reload" it is like this: shmem:total_size = 536870912 shmem:used_size = 28619016 shmem:real_used_size = 51842768 shmem:max_used_size = 76993616 shmem:free_size = 485028144 shmem:fragments = 722063 Notice the increase in fragments. Sequentials "kamctl cr reload" does not change the fragments allot. Any ideas? -- Morten Isaksen

3 4

maddr parameter in contact header and fix_nated_contact
by Morten Isaksen 12 Jun '12

12 Jun '12

Hi, I have this setup: Mediation server with internal ip -- NAT firewall -- Kamailio proxy -- PSTN gateway The mediation server sends a contact header with maddr=<internal IP> fix_nated_contact only changes the host part and leaves the maddr intact. When the PSTN gateway sends a PRACK or a BYE it puts the <internal IP> in the RURI, and kamailio then try to forward the message directly to the <internal IP> and not the NAT ip. >From RFC 3261 19.1.1: The maddr parameter indicates the server address to be contacted for this user, overriding any address derived from the host field. When an maddr parameter is present, the port and transport components of the URI apply to the address indicated in the maddr parameter value. [4] describes the proper interpretation of the transport, maddr, and hostport in order to obtain the destination address, port, and transport for sending a request. I can see that OpenSIPS has fixed this. http://sourceforge.net/tracker/?func=detail&atid=1086410&aid=3312423&group_… Would it be possible to get this patch into Kamailio also? -- Morten Isaksen

2 5

Install Kamailio v3.3.x from GIT
by Daniel-Constantin Mierla 12 Jun '12

12 Jun '12

Hello, now that the branch 3.3 is available, I created the tutorial about how to install Kamailio v3.3.x from GIT: * http://www.kamailio.org/wiki/install/3.3.x/git Feedback and improvements are welcome! 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 0

git branch 3.3 created
by Daniel-Constantin Mierla 11 Jun '12

11 Jun '12

Hello, GIT branch 3.3 was created to hold the release series 3.3.x -- version 3.3.0 will be out in max. 1 week, a decision after collecting eventual issues still open to be reported today. Here are few details about how to use it. Download and use branch 3.3 from git: git clone --depth 1 git://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 3.3 origin/3.3 Pushing patches: - only bug fixes or documentation improvements have to be committed to branch 3.3 - first commit to master branch (if the bug applies there as well), cherry-pick the commit to branch 3.3 and push to remote repository 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 0

development for future version 3.4.0 is open
by Daniel-Constantin Mierla 11 Jun '12

11 Jun '12

Hello, short note to let everyone know that new features can be committed again to the master branch -- they will be part of future version 3.4.0 (if no other numbering release will be decided). Testing phase of version 3.3.0 within master is ended, a dedicated branch being created for it (details in a separate message). Happy coding, 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 0

Capturing a users call (SIP+RTP+T.38)
by Klaus Darilion 11 Jun '12

11 Jun '12

Hi all! Do you know any comfortable tools to filter out a certain users call? e.g. searching in SIP packets for the user pattern, get the media port out of SDP and capture also RTP+T.38 I found pcapsipdump but have not tried it yet. Any suggestions or do I have to write my own tool? thanks Klaus

4 6

branching 3.3 - to be used for 3.3.x releases
by Daniel-Constantin Mierla 11 Jun '12

11 Jun '12

Hello, I am going to create the GIT branch 3.3 in short time -- this branch will be used for releasing 3.3.x versions. After that master can take again commits of new features, to be part of next major release 3.4.x. Branch 3.3 will take only fixes. From my point of view, I have two issues that I am trying to solve and the 3.3.0 is ready to go out, namely: - a solaris sigbus - pua_dialoginfo issue reported on mailing list Otherwise, nothing else critcal is out there, to my knowledge. Report them if you have other ones. Packaging specs were updated, so we are pretty much ready to release when no relevant issues is open. The wiki page with what is new in this release was updated a bit as well, add if you find something missing at: - http://www.kamailio.org/wiki/features/new-in-devel Also, contribute to the upgrade guidelines: - http://www.kamailio.org/wiki/install/upgrade/3.2.x-to-3.3.0 Cheer, 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

2 2

About kamailio 3.3 msrp module and sylkserver
by Min Wang 11 Jun '12

11 Jun '12

Hi I am wondering what are the major differences between kamailio 3.3 msrp module and sylkserver (http://sylkserver.com/features.phtml ) ? thanks. min

2 2

Dispatcher PSTN
by Kr0m 11 Jun '12

11 Jun '12

Hello I am not able to dial to pstn with kamailio, the call is routed to my pstn-gw(asterisk), but the final phone rings 4 or 5 seconds and then it is hanged up. My outbound route is: route[PSTN] { if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } route(TOASTERISK); exit; return; } route[TOASTERISK] { sl_send_reply("100","Trying"); uac_replace_from("$fn","sip:$fn@$fd"); route(NATMANAGE); ds_select_dst("1","4"); t_on_failure("1"); t_relay(); } failure_route[1] { ds_mark_dst("i"); if (!ds_next_dst()) { t_reply("503", "Service unavailable: no more dst"); exit; } route(TOASTERISK); } With a traffic capture i can see the traffic returning to my kamailio server. Any suggestion will be appreciated.

2 1

Kamailio and Nonce Count
by patrice.bodeven＠orange.com 11 Jun '12

11 Jun '12

Hello, I try to understand the nonce count handling. By parameter "modparam("auth", "nonce_expire", 21600)" I can define the validity of the Nonce in the time. But I want also to define the validity of the Nonce in the number of use. For exemple, I want to limit the reuse of the nonce at 64 times (including register, Invite) But I can see the definition of the nonce_count for all the SIP users on Kamailio and not user per user. If I understand when I use modparam("auth", "nonce_count", 1) # enable nonce_count support modparam("auth", "nc_array_order", 20) # 1M in-flight nonces, using 1Mb memory modparam("auth", "nid_pool_no", 4) I define a memory to store the nonce values for all the SIP users. So when User A use nonce x for REGISTER and nonce y for INVITE, nonce x is still valid ! My understanding is it correct ? Is anyway to define a nonce validity in number of times per user ? Patrice B _________________________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, France Telecom - Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, France Telecom - Orange is not liable for messages that have been modified, changed or falsified. Thank you.

2 1

Kamailio 3.2 + RTP proxy - no errors - no media
by Lukas Lani 10 Jun '12

10 Jun '12

Hi, I am trying to use RTPproxy 1.2.1 in bridge mode between IPv4 and IPv6 together with Kamailio 3.2. I configured and ran it on server together with Kamailio. I start it with parameters: -l 192.168.6.2 -6 /2001:6666:6666:6666::2 -s udp:localhost:7722 SIP messages are correctly exchanged, SDP looks to be translated, but when RTP session starts, I see RTP flow only from UC1 to RTPproxy and nothing more. On RTPproxy server I see 4 open ports for IPv4 address but no one for IPv6 address. When I tried to change start parameters to -6 2001:6666:6666:6666::2 -l /192.168.6.2 -s udp:localhost:7722 I could see 4 open ports for IPv6 address but no one for IPv4 address. Can anybody help? do you know about any known error? can it be related to Kamailio or RTPproxy? thank you, Lukas Lani

2 2

Looking for RTP Proxy in TCP
by Austin Einter 08 Jun '12

08 Jun '12

Hi All Now I am using Kamailio 3.1.5 and RTP proxy 1.1. Looks both are compatible and working fine. The RTP Proxy basically sends/receives RTP packets over UDP. Is there any RTP Proxy available that does send/receive of RTP packets over TCP and also should be compatible with Kamailio 3.1.5. If you have any information in this regard, kindly share. Thanks a lot. Austin

6 7

lcr from_uri not matching
by Ben WIlliams 08 Jun '12

08 Jun '12

Has anyone experienced this problem where lcr only works when from_uri is null. When I change it to .* the match fails. Thanks Ben

3 8

Re: [SR-Users] Issue using Kamailio as a proxy in front of Asterisk
by Daniel-Constantin Mierla 07 Jun '12

07 Jun '12

Do not write private emails, keep the mailing list cc-ed for the topics started there. Cheers, Daniel On 6/7/12 12:22 PM, Dominik Mauritz wrote: > Daniel, Stoyna, > > the solutions you described look very interesting to me. I will have a > look at both and pick the one that fits best. > > Thanks guys. > > Rgs, > Dominik > > > > Am 07.06.12 10:57, schrieb Daniel-Constantin Mierla: >> Hello, >> >> On 6/6/12 9:57 PM, Stoyan Mihaylov wrote: >>> We use Jitsi as SIP client, and openxcap along with camailio to >>> handle presence. Then jitsi know if account is online or offline. >>> Our Asterisk dont know nothing about accounts (it accepts all calls >>> from kamailio). There I run AGI scripts, which can check kamailio >>> tables - and I can know if account is online or offline. Of course >>> this do not work if account is "forced" to offline. >>> But may be there is better solution. >>> >>> On Wed, Jun 6, 2012 at 10:04 PM, Dominik Mauritz >>> <dominik.mauritz(a)web.de <mailto:dominik.mauritz@web.de>> wrote: >>> >>> I have already tried that. I defined SIP-Accounts in Asterisk >>> with host=<Kamailio-IP> (instead of host=dynamic). This solves the >>> described problem but it also has side effects: >>> >>> - You don't have the correct presence status on your phone (e. >>> g. xlite) indicating wether the account is online or offline >>> - Asterisk always sends invites to Kamailio on incoming calls >>> even if there is no phone registered to the account >>> >>> These are not the worst things in the world, but it is maybe not >>> the best solution possible. >>> >>> >>> >>> Am 06.06.12 20:41, schrieb Stoyan Mihaylov: >>> >>> We use also Kamailio in front of Asterisk - but I forward >>> only calls to Asterisk - register/unregister stay in Kamailio. >>> Asterisk dont know which device is registered, and which is not. >>> >>> >>> On Wed, Jun 6, 2012 at 8:20 PM, Dominik Mauritz >>> <dominik.mauritz(a)web.de <mailto:dominik.mauritz@web.de> >>> <mailto:dominik.mauritz@web.de <mailto:dominik.mauritz@web.de>>> wrote: >>> >>> Hi All, >>> >>> some days ago I installed Kamailio as a front end for >>> Asterisk following this tutorial: >>> >>> http://kb.asipto.com/asterisk:__realtime:kamailio-3.1.x-__asterisk-1.6.2-as… >>> <http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb> >>> >>> >>> >>> I added Multi Domain support and adjusted some other things >>> to fit my environment. Almost everything is working perfectly now. >>> One small issue is left: >>> >>> With Kamailio in front of Asterisk I have one feature (next >>> to other cool things) I was missing for years. I am now able to >>> register more than one device on the same SIP account. This is >>> nothing new for Kamailio users but Asterisk doesn’t support this. >>> >>> If I register two phones on one SIP account with Kamailio >>> everything is fine. They are able to make outgoing calls and both >>> ring on incoming calls. But if one phone unregisters Kamailio >>> forwards the unregister request to Asterisk and Asterisk sets this >>> account to offline. Now a call comes in but Asterisk is not sending >>> out an invite because for Asterisk the phone is offline. >>> >>> I wonder if it is possible to forward an unregister request >>> from Kamailio to Asterisk only if the last endpoint registered with >>> one SIP account unregisters. If there is more than one endpoint >>> using the same Account Kamailio should not forward the unregister >>> request to Asterisk. >>> >>> Any idea? >>> >> you can use reg_fetch_contacts(...) in your config to find out how >> many contacts are for an user and based on that do not send the >> un-register to asterisk, see: >> >> http://kamailio.org/docs/modules/stable/modules_k/registrar.html#id2498205 >> >> http://kamailio.org/docs/modules/stable/modules_k/registrar.html#id2498441 >> >> >> 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 >> >> >> > -- 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

2 1

record calls audio and video
by Saul Waizer 07 Jun '12

07 Jun '12

Hello List, I am looking for a way to record the audio and video of a call. I am using Kamailio (latest) on Ubuntu. Does anyone know if there's any existing modules that allow for recording or any other solution that can provide recording of calls on Kamailio? Thank you!

4 6

Siremis v3.2.1 Released
by Elena-Ramona Modroiu 07 Jun '12

07 Jun '12

Hi, Siremis v3.2.1 is out, the web management interface for Kamailio SIP Server. It is the last release with new features planned to be compatible with Kamailio v3.2.x. Next major releases for Siremis will target upcoming Kamailio v3.3.x. This time new features relate to web pages that allow management of own profile by SIP users, such as changing SIP password, management of speed dials or views for accounting, cdrs and location records. There was also added a form for doing user registrations (e.g., can be used for allowing public registrations). You can read more details about the release as well as details about installation at: * http://siremis.asipto.com/ Regards, Ramona

1 0

Limiting calls from users in prepaid scenerio
by Aft nix 07 Jun '12

07 Jun '12

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, We are thinking about launching a different "usage policy". The usage policy will be prepaid in some sense. but instead of limiting "number of minutes" we are thinking about limiting "number of calls". That means if you pay this X amount of doller, you can make Y amount of calls. calls can last whatever the user wants given that its under 1 hr. I've seen how to limit concurrent calls using SER, but the usage i have described, is not seen by me. So please help me out with this thing. My initial understanding about how it can be done : 1) when a dialog is confirmed with ACK(we have both callid, from tag, to tag), we can make an entry to the database by increasing "number of calls" field by one. 2) A field in database will be used as flag for usage limit. when "number of calls" exceeds the usage limit, it will flag it as "finished usage limit". 3) when a INVITE arrives, a check will be made with $fU against DB. With this framework in mind, I have read "db_mysql" module. And to my surprise, no direct the database functions are not exported to the cfg. I don't know if any module in kamailio exports mysql_queury() to cfg, which i need for my purpose. And may be there is some built in module which provides a higher level layer which will enable myself achieving my goal. If in the end i need "exported" mysql functions, then i can develop it myself. So which way i should go? Thanks in advance - -- - -aft -----BEGIN PGP SIGNATURE----- Version: OpenPGP.js v0.1 Comment: http://openpgpjs.org wsBcBAEBAgAQBQJPzc6dCRCJVJ6A/SK8awAARAkH/R5rqfiTPqBP+zZr8rgM 0JE42l9jxYq5OteD13U4CvcrUklXZlgDf2jwr1F7ndd1rlV3yKXAT2u61Bz5 LwNneRZydl3xnPmt2+TV6Qkcgy8RXgyRMntP9iLeQOXWJNW7D637h7tLL11d O50s86ofgJT+dfTICbomNZjLZRr1DvYOdcByG3LHcYx033G6bIFlMqJ0ZO+G cBdgtJcLrwo6iPn8Gv+5aJDy2kRUphrF7cVCzFzLPnNVZpvXNNcv7LB9bkUH cClcLDVAbsdw0TPd9Z6DNJSxFlHy2uzwNWB9mhpVJD1toT8Geb4iCOi+ZG8P WT5fgNwSoZlEm0BBBHwMUdk= =fcdD -----END PGP SIGNATURE-----

2 2

Issue using Kamailio as a proxy in front of Asterisk
by Dominik Mauritz 07 Jun '12

07 Jun '12

Hi All, some days ago I installed Kamailio as a front end for Asterisk following this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb I added Multi Domain support and adjusted some other things to fit my environment. Almost everything is working perfectly now. One small issue is left: With Kamailio in front of Asterisk I have one feature (next to other cool things) I was missing for years. I am now able to register more than one device on the same SIP account. This is nothing new for Kamailio users but Asterisk doesn’t support this. If I register two phones on one SIP account with Kamailio everything is fine. They are able to make outgoing calls and both ring on incoming calls. But if one phone unregisters Kamailio forwards the unregister request to Asterisk and Asterisk sets this account to offline. Now a call comes in but Asterisk is not sending out an invite because for Asterisk the phone is offline. I wonder if it is possible to forward an unregister request from Kamailio to Asterisk only if the last endpoint registered with one SIP account unregisters. If there is more than one endpoint using the same Account Kamailio should not forward the unregister request to Asterisk. Any idea? Thanks, Dominik

3 4

kamailio as SBC
by Mino Haluz 06 Jun '12

06 Jun '12

Hi, I know that kamailio is SIP proxy, but is there any way how to implement kamailio as SBC like OpenSIPS with B2BUA module ? I tried OpenSIPS with this module, but it does not work with mediaproxy module. Mino

5 4

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sr-users June 2012