sr-users January 2013

sr-users@lists.kamailio.org

82 participants
126 discussions

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

pkg mem problems. Leaking?
by Jon Bonilla 08 Apr '13

08 Apr '13

Hi all I have two different kamailio servers running in a single machine. The first one is a sl proxy and the second one acts as stateful proxy/registrar. From time to time the stateless one goes out of pkg mem. It has 8MB (also tried with 16 in another setup). The sl proxy listens the public interface for external connections and the loopback interface for conecting to the stateful proxy. As I have seen since I installed a mem debug kamailio 3.3.2 version there 6 days ago, the internal socket children have grown on mem usage while the external ones remain more or less stable. As example one uf the external udp receivers uses less than 1MB while one of the internal ones uses almost 6MB. I'm attaching the mem dump of one of those internal udp receiver processes via "ngcp-sercmd lb cfg.set_now_int core mem_dump_pkg 25437" command. I see several tens of thousands of lines like these ones: N address=0x7fd1207383c0 frag=0x7fd120738390 size=56 used=1 alloc'd from <core>: parser/parse_param.c: parse_params(539) start check=f0f0f0f0, end check= c0c0c0c0, abcdefed I have no idea about how to continue debugging this. I'm pasting some more info regarding the process list, mem status and kamailio version/flags. Thanks sp1:~# ngcp-sercmd lb ps 25416 attendant 25429 udp receiver child=0 sock=77.77.77.77:5060 25430 udp receiver child=1 sock=77.77.77.77:5060 25431 udp receiver child=2 sock=77.77.77.77:5060 25432 udp receiver child=3 sock=77.77.77.77:5060 25433 udp receiver child=4 sock=77.77.77.77:5060 25434 udp receiver child=5 sock=77.77.77.77:5060 25435 udp receiver child=6 sock=77.77.77.77:5060 25436 udp receiver child=7 sock=77.77.77.77:5060 25437 udp receiver child=0 sock=127.0.0.1:5060 25438 udp receiver child=1 sock=127.0.0.1:5060 25439 udp receiver child=2 sock=127.0.0.1:5060 25440 udp receiver child=3 sock=127.0.0.1:5060 25441 udp receiver child=4 sock=127.0.0.1:5060 25442 udp receiver child=5 sock=127.0.0.1:5060 25443 udp receiver child=6 sock=127.0.0.1:5060 25444 udp receiver child=7 sock=127.0.0.1:5060 25445 slow timer 25446 timer 25447 ctl handler 25448 MI FIFO 25449 tcp receiver (generic) child=0 25450 tcp receiver (generic) child=1 25451 tcp receiver (generic) child=2 25452 tcp receiver (generic) child=3 25453 tcp receiver (generic) child=4 25454 tcp receiver (generic) child=5 25455 tcp receiver (generic) child=6 25456 tcp receiver (generic) child=7 25457 tcp receiver (tcp:127.0.0.1:5060) child=8 25458 tcp receiver (tcp:127.0.0.1:5060) child=9 25459 tcp main process sp1:~# ngcp-sercmd lb pkg.stats { entry: 0 pid: 25416 rank: 0 used: 901136 free: 15417544 real_used: 1359672 } { entry: 1 pid: 25429 rank: 1 used: 909360 free: 15408648 real_used: 1368568 } { entry: 2 pid: 25430 rank: 2 used: 909400 free: 15408608 real_used: 1368608 } { entry: 3 pid: 25431 rank: 3 used: 909352 free: 15408656 real_used: 1368560 } { entry: 4 pid: 25432 rank: 4 used: 909344 free: 15408664 real_used: 1368552 } { entry: 5 pid: 25433 rank: 5 used: 909392 free: 15408712 real_used: 1368504 } { entry: 6 pid: 25434 rank: 6 used: 909344 free: 15408664 real_used: 1368552 } { entry: 7 pid: 25435 rank: 7 used: 909344 free: 15408664 real_used: 1368552 } { entry: 8 pid: 25436 rank: 8 used: 909352 free: 15408656 real_used: 1368560 } { entry: 9 pid: 25437 rank: 9 used: 5884872 free: 2480784 real_used: 14296432 } { entry: 10 pid: 25438 rank: 10 used: 5935272 free: 2359728 real_used: 14417488 } { entry: 11 pid: 25439 rank: 11 used: 5971616 free: 2305912 real_used: 14471304 } { entry: 12 pid: 25440 rank: 12 used: 5821672 free: 2522672 real_used: 14254544 } { entry: 13 pid: 25441 rank: 13 used: 5945584 free: 2310056 real_used: 14467160 } { entry: 14 pid: 25442 rank: 14 used: 5868760 free: 2495264 real_used: 14281952 } { entry: 15 pid: 25443 rank: 15 used: 5903672 free: 2457952 real_used: 14319264 } { entry: 16 pid: 25444 rank: 16 used: 5899208 free: 2453872 real_used: 14323344 } { entry: 17 pid: 25445 rank: -1 used: 983944 free: 15409512 real_used: 2267696 } { entry: 18 pid: 25446 rank: -1 used: 909168 free: 15409512 real_used: 1367704 } { entry: 19 pid: 25447 rank: -2 used: 983664 free: 15407912 real_used: 2267032 } { entry: 20 pid: 0 rank: 0 used: 918424 free: 15400064 real_used: 1377152 } { entry: 21 pid: 25449 rank: 17 used: 989192 free: 15329200 real_used: 1448016 } { entry: 22 pid: 25450 rank: 18 used: 989192 free: 15329200 real_used: 1448016 } { entry: 23 pid: 25451 rank: 19 used: 989192 free: 15329200 real_used: 1448016 } { entry: 24 pid: 25452 rank: 20 used: 989192 free: 15329200 real_used: 1448016 } { entry: 25 pid: 25453 rank: 21 used: 989192 free: 15329200 real_used: 1448016 } { entry: 26 pid: 25454 rank: 22 used: 989192 free: 15329200 real_used: 1448016 } { entry: 27 pid: 25455 rank: 23 used: 989192 free: 15329200 real_used: 1448016 } { entry: 28 pid: 25456 rank: 24 used: 989192 free: 15329200 real_used: 1448016 } { entry: 29 pid: 25457 rank: 25 used: 989192 free: 15329200 real_used: 1448016 } { entry: 30 pid: 25458 rank: 26 used: 989192 free: 15329200 real_used: 1448016 } { entry: 31 pid: 25459 rank: -4 used: 983240 free: 15334096 real_used: 2266992 } sp1:~# kamailio -V version: kamailio 3.3.2 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 13:08:06 Oct 31 2012 with gcc 4.4.5

3 3

Debugging and syslog
by Daniel W. Graham 26 Mar '13

26 Mar '13

Hello, Reaching out for some assistance with debugging, I am not seeing any xlog messages in my log file. If there is any other info I didn't include that would be helpful please let me know. Any ideas are appreciated. Using Kamailio 3.3.2 & Cent OS 6 Rsyslog.conf -------------------- *.info;mail.none;authpriv.none;cron.none;local0.none /var/log/messages # Kamailio local0.* -/var/log/kamailio Kamailio CFG ------------------------------ #!define WITH_DEBUG #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif log_facility=LOG_LOCAL0 loadmodule "xlog.so" Example xlog- xlog("L_INFO", "No more gateways \n"); Thanks, Dan-

5 10

Need help understanding/separating signaling from media
by Grant Bagdasarian 13 Mar '13

13 Mar '13

Hello, I want to separate signaling and media in our architecture but I'm not entirely sure if the following will work (note that it is still in a design phase and these kinds of constructs are still new to me): User Agent 1-------------- Kamailio -------------- User Agent 2 | | | | | | -------------- Media Server Cluster -------------- The signaling will go through Kamailio but the media stream should pass through one of the media servers in the cluster for each call. Is it possible to achieve this by having Kamailio instruct the media server which ports to open and alter sdp information? I think that the RTP Proxy module does precisely this, but what I still don't understand is how the rtp stream is passed between these two agents for each call. Also, is RTP proxy the best way to do this or are there any other third party media servers which I could use? I don't need any fancy functionality, just basic media handling/pass-through. Regards, Grant

2 8

Send Bye to all calls in one profile
by Camila Troncoso 13 Mar '13

13 Mar '13

Hi, I have a kamailio proxy server that´s controlling the limit of calls that clients can make with profile grouping. In other side I also limit clients call duration base on users credit ( DATABASE) and *timeout *feature . Before the calls get answer I calculate how much time the client has left for this call and set the timeout variable. When the time is reached the proxy sends BYEs two both sides of this call. I want when the call timeout, as before explain, all other calls in the same profile end too, because the client has expired his credit. Is there a way to achieve this? Regards, Camila *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

4 6

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

why use received as RURI for nathelper OPTIONS ping?
by Andrew Pogrebennyk 11 Mar '13

11 Mar '13

Some Fritzbox devices don't accept the OPTIONS ping generated by kamailio with RURI: sip:11.22.3.4:5060. In the location table we have: received: sip:11.22.3.4:5060 contact: sip:user@11.22.3.4:5060;uniq=6633BC1386F4D4CC4EBD64DC7E967 path: <sip:lb@127.0.0.1;lr;received='sip:11.22.3.4:5060'> Kamailio version is 3.3.2 and the nathelper config is nothing fancy: modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@sipwise.local") modparam("nathelper", "received_avp", "$avp(s:received)") What I don't understand is why kamailio sets RURI of the OPTIONS message to value of "received" instead of the contact. I suspect a bug in the parser somewhere along these lines: > rval = ul.get_all_ucontacts(buf,cblen,(ping_nated_only?ul.nat_flag:0), > ((unsigned int)(unsigned long)timer_idx)*natping_interval+iteration, > natping_processes*natping_interval); > if (rval != 0) { > pkg_free(buf); > goto done; > } > } > > if (buf == NULL) > goto done; > > cp = buf; > while (1) { > memcpy(&(c.len), cp, sizeof(c.len)); > if (c.len == 0) > break; > c.s = (char*)cp + sizeof(c.len); > cp = (char*)cp + sizeof(c.len) + c.len; > memcpy( &send_sock, cp, sizeof(send_sock)); > cp = (char*)cp + sizeof(send_sock); > memcpy( &flags, cp, sizeof(flags)); > cp = (char*)cp + sizeof(flags); > memcpy( &(path.len), cp, sizeof(path.len)); > path.s = path.len ? ((char*)cp + sizeof(path.len)) : NULL ; > cp = (char*)cp + sizeof(path.len) + path.len; > > /* determin the destination */ > if ( path.len && (flags&sipping_flag)!=0 ) { > /* send to first URI in path */ > if (get_path_dst_uri( &path, &opt) < 0) { > LM_ERR("failed to get dst_uri for Path\n"); > continue; > } > /* send to the contact/received */ > if (parse_uri(opt.s, opt.len, &curi) < 0) { > LM_ERR("can't parse contact dst_uri\n"); > continue; > } > } else {

4 5

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.

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Help with Asterisk RT integration
by Jon Morby 28 Feb '13

28 Feb '13

Hi I'm trying to integrate a (K) front end cluster with an Asterisk back end cluster and Asterisk RT (legacy system) I've followed the recipe at http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb (with one minor exception …) if (from_uri!=myself && uri!=myself) became if (uri!=myself) as with the original line in place we were able to spoof traffic from 3rd party sites and route out onto the PSTN (which I thought was bad) … anyway … The problem I'm seeing currently is that when a call is passed down a SIP trunk to an end user on the (K) platform we're losing the DNID Asterisk delivers the call to SIP/account/DNID (K) however just tries to deliver the call to DNID@domain which comes back as not found. I've had limited success hand coding aliases into the alias_db however we're still missing the DNID info and having to scrape it from the SIP To field (with a) limited success and b) fears of a support nightmare if we try and move existing customers onto the new platform). If we were to do this for real I'd either have to modify usrloc or try some perl_exec magic I'm guessing … (although it might be possible with a view on the existing sql database … but I'd rather not have to do that if there is a simpler way) I confess I'm struggling to get my head around the config and docs and hoped someone could point me in the right direction? For legacy reasons Asterisk needs to be in the critical path on this particular build … what I'm looking for is a simple recipe and some helpful pointers on how to implement it that will allow enable me to swing (K) into the path between our end user SIP devices and the existing asterisk back ends without losing the ability to deliver hundreds of numbers down a single SIP trunk to a subscriber, and that doesn't require them to make any changes on their end as they will still see the equivalent of SIP:${DNID}@example.com arriving on their PBX This should be simple, but I'm obviously missing something :) Help and pointers gratefully received -- Jon Morby FidoNet - the internet made simple! 10 - 16 Tiller Road, London, E14 8PX tel: 0845 004 3050 / fax: 0845 004 3051

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sr-users January 2013