- sr-users - kamailio.org

$time(year) gives 1900
by Uri Shacked 20 Feb '12

20 Feb '12

Hi, I am using the $time(name) pv. On my lab machine (centos 5.7) i get the right value (2012). But, on my other machine (centos 6.2) i get 1900. what am i missing here?

2 2

Mtree VS Htable getting huge amount of data from DB not on local machine
by Uri Shacked 20 Feb '12

20 Feb '12

> > Hi, > > untill today i used the mtree module to get some data from a remote DB and > load it into memory. > the table in the data base has 5 million rows. prefix is a number and > value is some string. > > i moved to htable because i need the prefix exact match and not longest.... > > now, after 5-10 sec, i get the following error (that i never got in mtree) > - "lost connection to mysql during query" and kamailio does not start. > > any ideas how to solve it? > > cheers, > > Uri >

2 2

Registration replication
by Spencer Thomason 19 Feb '12

19 Feb '12

Hello, I'm currently working on a HA setup with geographically separated proxies. Each proxy is in a different data center with an MPLS link between them. The proxies access a locally connected PostgreSQL database that is replicated from a master in the primary data center. Note that the replicated dbs are read only. Everything works great for read only queries but registrations seem to pose a problem as they must use the primary db directly to deal with the writes. If I set userloc to db mode 2 the secondary proxies wipe out the entries in the database. If I use db mode 3 everything works great but all of the secondary proxies must use the primary database directly and a network link failure will cause a service disruption. Is there any way I can use separate read and write databases without p_userloc? Is there a better way to replicate registrations maintaining availability and scalability? And advice is appreciated. Thanks, Spencer Connected by DROID on Verizon Wireless

2 1

Segmentation Fault encountered in pv.so
by Simpson Chua 19 Feb '12

19 Feb '12

Hi, Since I am getting errors in from the usrloc module, I am trying another way to parse out the information I need using pv.so. I'm encountering similar segfaults issues. Here are snippets of my configs: modparam("sipcapture", "db_url", "mysql://xxx:xxx@localhost/test_db") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "raw_socket_listen", "127.0.0.1:5060-9000") modparam("sipcapture", "raw_interface", "eth2") modparam("sipcapture", "raw_sock_children", 4) modparam("sipcapture", "raw_moni_capture_on", 1) modparam("sipcapture", "promiscious_on", 1) route { if (method == "REGISTER") { if (avp_check("$si","eq/10.xxx.xxx.xxx")) { #save("location", "0x04"); # This currently results in segfault and is being addressed by Alexandr sip_capture(); xlog("The received IP is $Ri\n"); # Just to test if I am able to parse the received IP from the Via headers correctly drop; } drop; } } Feb 17 13:41:02 ubuntu kernel: [251382.760104] kamailio[9414]: segfault at 31 ip 00007fb4ef077611 sp 00007fff0c8a6468 error 4 in pv.so[7fb4ef06a000+39000] Core was generated by `kamailio -w /home/core'. Program terminated with signal 11, Segmentation fault. #0 pv_get_rcvip (msg=0x7fb4efdd7598, param=0x7fb4efdcef88, res=0x7fff0c8a64b0) at pv_core.c:647 647 || msg->rcv.bind_address->address_str.s==NULL) (gdb) bt #0 pv_get_rcvip (msg=0x7fb4efdd7598, param=0x7fb4efdcef88, res=0x7fff0c8a64b0) at pv_core.c:647 #1 0x000000000049b5ad in pv_get_spec_value (msg=0x7fb4efdd7598, sp=0x7fb4efdcef70, value=0x7fff0c8a64b0) at pvapi.c:1180 #2 0x000000000049b71d in pv_printf (msg=0x7fb4efdd7598, list=<optimized out>, buf=<optimized out>, len=0x7fff0c8a6558) at pvapi.c:1239 #3 0x00007fb4ed0d442e in xlog_helper (msg=<optimized out>, xm=0x7fb4efdceef0, level=-1, line=0) at xlog.c:168 #4 0x000000000041babb in do_action (h=0x7fff0c8a73b0, a=0x7fb4efdd02a0, msg=0x7fb4efdd7598) at action.c:1122 #5 0x000000000041abe2 in run_actions (h=0x7fff0c8a73b0, a=0x7fb4efdd0080, msg=0x7fb4efdd7598) at action.c:1610 #6 0x000000000041c05b in do_action (h=0x7fff0c8a73b0, a=0x7fb4efdd10d0, msg=0x7fb4efdd7598) at action.c:1107 #7 0x000000000041abe2 in run_actions (h=0x7fff0c8a73b0, a=0x7fb4efdd10d0, msg=0x7fb4efdd7598) at action.c:1610 #8 0x000000000041c05b in do_action (h=0x7fff0c8a73b0, a=0x7fb4efdd1350, msg=0x7fb4efdd7598) at action.c:1107 #9 0x000000000041abe2 in run_actions (h=0x7fff0c8a73b0, a=0x7fb4efdd1350, msg=0x7fb4efdd7598) at action.c:1610 #10 0x0000000000422fa2 in run_top_route (a=0x7fb4efdd1350, msg=0x7fb4efdd7598, c=<optimized out>) at action.c:1683 #11 0x00000000004b061e in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:207 #12 0x00007fb4edb538b0 in raw_capture_rcv_loop (rsock=5, port1=5060, port2=8933, ipip=0) at sipcapture.c:1518 #13 0x00007fb4edb53e1f in init_rawsock_children () at sipcapture.c:572 #14 0x00007fb4edb54033 in child_init (rank=<optimized out>) at sipcapture.c:542 #15 0x0000000000507a0f in init_mod_child (m=0x7fb4efdcb2b0, rank=0) at sr_module.c:886 #16 0x0000000000507974 in init_mod_child (m=0x7fb4efdcb668, rank=0) at sr_module.c:883 #17 0x0000000000507974 in init_mod_child (m=0x7fb4efdcc178, rank=0) at sr_module.c:883 #18 0x0000000000507974 in init_mod_child (m=0x7fb4efdcc4b0, rank=0) at sr_module.c:883 #19 0x0000000000507974 in init_mod_child (m=0x7fb4efdccaf0, rank=0) at sr_module.c:883 #20 0x0000000000507974 in init_mod_child (m=0x7fb4efdcd030, rank=0) at sr_module.c:883 #21 0x0000000000476149 in main_loop () at main.c:1661 #22 0x000000000041a940 in main (argc=<optimized out>, argv=0x7fff0c8a7be8) at main.c:2475 (gdb) p *msg $1 = {id = 82, pid = 9418, fwd_send_flags = {f = 0 '\000', blst_imask = 0 '\000'}, rpl_send_flags = {f = 0 '\000', blst_imask = 0 '\000'}, first_line = { type = 1, len = 43, u = {request = {method = { s = 0x7fb4edd57d2a "REGISTER sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCal"..., len = 8}, uri = { s = 0x7fb4edd57d33 "sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 225"..., len = 24}, version = { s = 0x7fb4edd57d4c "SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 2257e150-b49dfd8d-c8d924b2@1"..., len = 7}, method_value = 32}, reply = {version = { s = 0x7fb4edd57d2a "REGISTER sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCal"..., len = 8}, status = { s = 0x7fb4edd57d33 "sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 225"..., len = 24}, reason = { s = 0x7fb4edd57d4c "SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 2257e150-b49dfd8d-c8d924b2@1"..., len = 7}, statuscode = 32}}}, via1 = 0x7fb4efdd7ef0, via2 = 0x7fb4efdd8218, headers = 0x7fb4efdd7e50, last_header = 0x7fb4efdd87c0, parsed_flag = 18446744073709551615, h_via1 = 0x7fb4efdd80d8, h_via2 = 0x7fb4efdd8178, callid = 0x7fb4efdd19b0, to = 0x7fb4efdd1910, cseq = 0x7fb4efdd7c80, from = 0x7fb4efdd7e50, contact = 0x7fb4efdd84a0, maxforwards = 0x7fb4efdd8680, route = 0x0, record_route = 0x0, content_type = 0x0, content_length = 0x7fb4efdd87c0, authorization = 0x0, expires = 0x7fb4efdd8720, proxy_auth = 0x0, supported = 0x0, require = 0x0, proxy_require = 0x0, unsupported = 0x0, allow = 0x0, event = 0x0, accept = 0x0, accept_language = 0x7fb4efdd85e0, organization = 0x0, priority = 0x0, subject = 0x0, user_agent = 0x7fb4efdd8540, server = 0x0, content_disposition = 0x0, accept_disposition = 0x0, diversion = 0x0, rpid = 0x0, refer_to = 0x0, session_expires = 0x0, min_se = 0x0, sipifmatch = 0x0, subscription_state = 0x0, date = 0x0, identity = 0x0, identity_info = 0x0, pai = 0x0, ppi = 0x0, path = 0x0, privacy = 0x0, body = 0x0, eoh = 0x7fb4edd58036 "12963462\", realm=\"proxyApp\", nonce=\"proxyAppXgyrmj82aTujp3sdBW\", uri=\"sip:abc.def.com\", response=\"039891b3f5999b90a645d3e808c2147c\", algorithm=MD5, cnonce=\"l7/u88HtGJCOAmM\", qop=auth, nc="..., unparsed = 0x7fb4edd58036 "12963462\", realm=\"proxyApp\", nonce=\"proxyAppXgyrmj82aTujp3sdBW\", uri=\"sip:abc.def.com\", response=\"039891b3f5999b90a645d3e808c2147c\", algorithm=MD5, cnonce=\"l7/u88HtGJCOAmM\", qop=auth, nc="..., rcv = {src_ip = {af = 2, len = 4, u = {addrl = {140411255757578, 4294967291}, addr32 = {184916746, 32692, 4294967291, 0}, addr16 = {39690, 2821, 32692, 0, 65531, 65535, 0, 0}, addr = "\n\233\005\v\264\177\000\000\373\377\377\377\000\000\000"}}, dst_ip = {af = 2, len = 4, u = {addrl = {140414021193994, 5273972}, addr32 = { 2950353162, 32692, 5273972, 0}, addr16 = {53514, 45018, 32692, 0, 31092, 80, 0, 0}, addr = "\n\321Ú¯\264\177\000\000tyP\000\000\000\000"}}, src_port = 5060, dst_port = 5060, proto_reserved1 = 32692, proto_reserved2 = -266587208, src_su = {s = {sa_family = 2, sa_data = "\023\304\n\233\005\v\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 184916746}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 184916746, sin6_addr = {__in6_u = { __u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x1, proto = 1 '\001'}, buf = 0x7fb4edd57d2a "REGISTER sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCal"..., len = 782, new_uri = {s = 0x0, len = 0}, dst_uri = {s = 0x0, len = 0}, parsed_uri_ok = 1, parsed_uri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = { s = 0x7fb4edd57d37 "abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 2257e15"..., len = 20}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 0, proto = 0, type = SIP_URI_T, flags = 0, transport = {s = 0x0, len = 0}, ttl = { s = 0x0, len = 0}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, transport_val = {s = 0x0, len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}}, parsed_orig_ruri_ok = 0, parsed_orig_ruri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x0, len = 0}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = 0, transport = {s = 0x0, len = 0}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, transport_val = {s = 0x0, len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = { s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}}, add_rm = 0x0, body_lumps = 0x0, reply_lump = 0x0, add_to_branch_s = '\000' <repeats 57 times>, add_to_branch_len = 0, hash_index = 0, msg_flags = 0, flags = 0, set_global_address = {s = 0x0, len = 0}, set_global_port = {s = 0x0, len = 0}, force_send_socket = 0x0, path_vec = {s = 0x0, len = 0}} (gdb) p *msg->rcv.bind_address Cannot access memory at address 0x1 (gdb) Thanks, Simpson ________________________________ From: "sr-users-request(a)lists.sip-router.org" <sr-users-request(a)lists.sip-router.org> To: sr-users(a)lists.sip-router.org Sent: Friday, February 17, 2012 7:34 AM Subject: sr-users Digest, Vol 81, Issue 52 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: ndb_redis module fails after a while (Daniel-Constantin Mierla) 2. Re: Kamailio Exits With: Segmentation Fault Error 4 in usrloc.so (Alexandr Dubovikov) ---------------------------------------------------------------------- Message: 1 Date: Fri, 17 Feb 2012 14:30:44 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] ndb_redis module fails after a while To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Cc: Javier Gallart <jgallartm(a)gmail.com> Message-ID: <4F3E5684.3090203(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Hello, right, very dummy paste position for resetting the redis context, use the attached patch or exchange lines 228 and 229. in redis_client.c Thanks, Daniel On 2/17/12 1:13 PM, Javier Gallart wrote: > Hi Daniel, Andrew > > I've just tested the patch and kamailio crashes when the redis server > is stopped. This is what I could get: > -From the logs: > Feb 17 06:33:32 r-gate-test ./kamailio[23812]: ALERT: <core> > [main.c:751]: child process 23819 exited by a signal 11 > Feb 17 06:33:32 r-gate-test ./kamailio[23812]: ALERT: <core> > [main.c:754]: core was generated > Feb 17 06:33:32 r-gate-test ./kamailio[23812]: INFO: <core> > [main.c:766]: INFO: terminating due to SIGCHLD > > -An the backtrace: > (...) > Core was generated by `./kamailio -f ../etc/kamailio/kamailio.cfg'. > Program terminated with signal 11, Segmentation fault. > #0 redisFree (c=0x0) at hiredis.c:817 > 817 if (c->fd > 0) > (gdb) bt > #0 redisFree (c=0x0) at hiredis.c:817 > #1 0x00007f726f035461 in redisc_reconnect_server > (rsrv=0x7f7271c93ac0) at redis_client.c:229 > #2 0x00007f726f037240 in redisc_exec (srv=<value optimized out>, > cmd=0x7fffe98c6090, argv1=<value optimized out>, argv2=<value > optimized out>, argv3=<value optimized out>, > res=<value optimized out>) at redis_client.c:298 > #3 0x00007f726f034f7d in w_redis_cmd3 (msg=0x7f7271d7b018, > ssrv=<value optimized out>, scmd=<value optimized out>, > sres=0x7f7271d74b58 "h\341\313qr\177") at ndb_redis_mod.c:156 > #4 0x0000000000417025 in do_action (h=0x7fffe98c6570, > a=0x7f7271cc5248, msg=<value optimized out>) at action.c:1134 > #5 0x000000000041e58b in run_actions (h=<value optimized out>, > a=<value optimized out>, msg=<value optimized out>) at action.c:1610 > #6 0x000000000041e8d4 in run_actions_safe (h=0x7fffe98c7610, > a=0x7f7271c93781, msg=0x7f7271c93780) at action.c:1662 > #7 0x00000000004b731d in rval_get_int (h=0x7fffe98c7610, msg=0x0, > i=0x7fffe98c6bd8, rv=0x3, cache=0x4) at rvalue.c:920 > #8 0x00000000004bb87c in rval_expr_eval_int (h=0x7fffe98c7610, > msg=0x7f7271d7b018, res=0x7fffe98c6bd8, rve=0x7f7271cc6768) at > rvalue.c:1914 > #9 0x0000000000417c7c in do_action (h=0x7fffe98c7610, > a=0x7f7271ccb9d0, msg=<value optimized out>) at action.c:1092 > #10 0x000000000041e58b in run_actions (h=<value optimized out>, > a=<value optimized out>, msg=<value optimized out>) at action.c:1610 > #11 0x0000000000417cd7 in do_action (h=0x7fffe98c7610, > a=0x7f7271ccbb10, msg=<value optimized out>) at action.c:1111 > #12 0x000000000041e58b in run_actions (h=<value optimized out>, > a=<value optimized out>, msg=<value optimized out>) at action.c:1610 > #13 0x000000000041795e in do_action (h=0x7fffe98c7610, a=<value > optimized out>, msg=<value optimized out>) at action.c:732 > #14 0x000000000041e58b in run_actions (h=<value optimized out>, > a=<value optimized out>, msg=<value optimized out>) at action.c:1610 > #15 0x000000000041e862 in run_top_route (a=0x7f7271c94888, > msg=0x7f7271d7b018, c=<value optimized out>) at action.c:1683 > #16 0x0000000000498f36 in receive_msg ( > buf=0x8bb100 "INVITE sip:34661574758@79.170.68.215:5060 > <http://sip:34661574758@79.170.68.215:5060> SIP/2.0\r\nVia: > SIP/2.0/UDP 79.170.68.214:5060;branch=z9hG4bK-9451-1-0\r\nFrom: > 34661574758 <sip:34661574758@79.170.68.214:5060 > <http://sip:34661574758@79.170.68.214:5060>>;tag=9451SIPpTag001\r\nTo: > sut <sip:"..., len=<value optimized out>, rcv_info=0x7fffe98c7910) at > receive.c:207 > #17 0x0000000000525987 in udp_rcv_loop () at udp_server.c:544 > #18 0x00000000004635f4 in main_loop () at main.c:1585 > #19 0x0000000000465e62 in main (argc=3, argv=0x7fffe98c7c08) at > main.c:2475 > > > Regards > > Javi > > On Fri, Feb 17, 2012 at 11:39 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > thanks for testing, indeed it was an extra declaration left over. > Can you try with the new patch attached? > > Cheers, > Daniel > > > On 2/17/12 11:10 AM, Andrew Pogrebennyk wrote: > > Hi Daniel, > > On 02/17/2012 10:47 AM, Daniel-Constantin Mierla wrote: > > I made a patch for server reconnect -- I had no access to > a computer > with redis lib installed for the moment, hopefully it > compiles. If you > can try and tell the result, it would be great, I can > commit then. > > I may be able to test this patch as well. Currently > compilations bails > out on attempt to redeclare redisc_reconnect_server function > parameter: > > CC (gcc) [M ndb_redis.so] ndb_redis_mod.o > CC (gcc) [M ndb_redis.so] redis_client.o > redis_client.c: In function 'redisc_reconnect_server': > redis_client.c:206:19: error: 'rsrv' redeclared as different > kind of symbol > redis_client.c:202:46: note: previous definition of 'rsrv' was > here > make[1]: *** [redis_client.o] Error 1 > make: *** [modules] Error 1 > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla -- http://www.asipto.com > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > > _______________________________________________ > 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://linkedin.com/in/miconda -- http://twitter.com/miconda

2 1

avp_db_query - closing the connection
by graham 19 Feb '12

19 Feb '12

Hi all, I have been using more and more avps for per-call queries populated by custom mysql queries. I would like to know how the connections of mysql queries are being closed. I tried to locate a command to do such, but unable to locate. Do they close on their own, after a certain timeout period? I am running kamailio 1.5.5, notls. The proxies are getting busier and I want to keep all the resources in check ... Thanks, -graham

2 1

question on getting RTP stats
by Nelson Pereira 19 Feb '12

19 Feb '12

Is there a way to get asterisk to publish RTP stats in the bye message? I would like to parse theses to DB and get stats from them. If so, how? Thanks Np

1 0

Kamailio or VOS3000
by Abid Naseem 17 Feb '12

17 Feb '12

How kamailio is different from VOS3000 or Nextone or PORTASwitch when we are talking about wholesale routing? Sent from my iPhone

2 1

Re: [SR-Users] Need example of Kamailio.cfg for a port mirror setup
by Nelson Pereira 17 Feb '12

17 Feb '12

I have libpcap, it's in /usr/lib64 but it's libpcap0.9.8 >:/usr/local/src/captagent # find / |grep libpcap /usr/share/doc/packages/libpcap-devel /usr/share/doc/packages/libpcap-devel/pcap.xml /usr/share/doc/packages/libpcap-devel/pcap.html /usr/share/doc/packages/libpcap-devel/pcap.txt /usr/share/doc/packages/libpcap0 /usr/share/doc/packages/libpcap0/LICENSE /usr/share/doc/packages/libpcap0/TODO /usr/share/doc/packages/libpcap0/CREDITS /usr/share/doc/packages/libpcap0/README.linux /usr/share/doc/packages/libpcap0/README /usr/share/doc/packages/libpcap0/CHANGES /usr/lib64/libpcap.a /usr/lib64/libpcap.so.0.9.8 /usr/lib64/libpcap.so /usr/lib64/libpcap.so.0 On Fri, Feb 17, 2012 at 10:38 AM, Max Mühlbronner <mm(a)42com.com> wrote: > This is a multi-part message in MIME format. > --------------070003040807050803020008 > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > Content-Transfer-Encoding: 8bit > > Hi, > > you are missing libpcap0.8 which is a dependency for the captagent. > > > BR > > Max M. > > Am 17.02.2012 16:31, schrieb Nelson Pereira: > > ok, tried to do the make install, getting the bellow: > > aurast:/usr/local/src/captagent # /usr/local/bin/captagent > > /usr/local/bin/captagent: error while loading shared libraries: > > libpcap.so.0.8: cannot open shared object file: No such file or directory > > > > Do i need to compile this? > > > > Also just tried running the captagent: > > aurast:/usr/local/src/captagent # ./captagent > > ./captagent: error while loading shared libraries: libpcap.so.0.8: > > cannot open shared object file: No such file or directory > > > > What am I doing wrong? > > > > This is a SUSE SLES11 64bit server running asterisk 1.8.6 > > > > Thanks > > > > > > > > On Fri, Feb 17, 2012 at 10:24 AM, Nelson Pereira <kitkat0981(a)gmail.com > > <mailto:kitkat0981@gmail.com>> wrote: > > > > from the atttached kamailio.cfg file, what should be changed to > > get captagent receiver working in kamailio instead of port mirroring? > > > > Thanks > > > > > > On Fri, Feb 17, 2012 at 7:29 AM, Alexandr Dubovikov > > <voip(a)start4.info <mailto:voip@start4.info>> wrote: > > > > > > The link has been not changed, you know it already: > > > > http://code.google.com/p/homer/wiki/HOWTO > > > > Homer Capture Agent (for all other unsupported system) > > > > if your SIP system (PBX, proxy) doesn't have a built-in > > capture agent with HEP or IPIP encapsulation support, you can > > use Homer Capture Agent (captagent). This is an external > > application (daemon) that collect SIP messages thru PCAPlib > > and send them to your capture server. The captagent support > > IPv4 and IPv6. > > > > preventing your next question: the captagent your can find in > > the git. > > > > http://code.google.com/p/homer/source/browse/ > > > > > > Wbr, > > Alexandr > > > > > > > > 17.02.2012 13:15, Nelson Pereira wrote: > >> Can captagent be installed on a asterisk server? If so, is > >> there a howto? > >> > >> Sent from my iPhone4 on IOS4 > >> > >> On Feb 17, 2012, at 7:07 AM, Alexandr Dubovikov > >> <voip(a)start4.info <mailto:voip@start4.info>> wrote: > >> > >>> Hi, > >>> > >>> start two captagents instance, each on own interface and > >>> duplicate all traffic to your kamailio node. (HEP protocol) > >>> > >>> Wbr, > >>> Alexandr > >>> > >>> 16.02.2012 17:51, Nelson Pereira wrote: > >>>> I would like to know if someone can show me an example of > >>>> there setup configuration in kamailio.cfg for capturing > >>>> packets comming in from a mirror port. > >>>> The reason I ask is that if I do a TCPDUMP on the server, I > >>>> see all traffic to and from different sources on my Voice > >>>> network. > >>>> > >>>> But when I use Homer SIPCAPTURE which uses Kamailio in the > >>>> background, I only see traffic sent FROM my PBX but not > >>>> seing the traffic comming back to it. > >>>> > >>>> I mean I only see one side of the call, yet TCPDUMP > >>>> confirmed I see everything.... > >>>> > >>>> What am I missing ? > >>>> > >>>> Thanks > >>>> > >>>> NP > >>>> > >>>> > >>>> _______________________________________________ > >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > >>>> sr-users(a)lists.sip-router.org <mailto: > sr-users(a)lists.sip-router.org> > >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >>> > >>> _______________________________________________ > >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > >>> mailing list > >>> sr-users(a)lists.sip-router.org > >>> <mailto:sr-users@lists.sip-router.org> > >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >> > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > >> sr-users(a)lists.sip-router.org <mailto: > sr-users(a)lists.sip-router.org> > >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > > mailing list > > sr-users(a)lists.sip-router.org > > <mailto:sr-users@lists.sip-router.org> > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > > > > > _______________________________________________ > > 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 > > > -- > 42com Telecommunication GmbH > Straße der Pariser Kommune 12-16 / D-10243 Berlin > > E-mail: mm(a)42com.com > Homepage: www.42com.com > > Firmenangaben/Company information: > > Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B > Umsatzsteuer-ID/VAT-ID: DE223812306, Geschäftsführer/CEO: Thomas Reinig, > Alexander Reinig > > > Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. Diese > sind möglicherweise vertraulich und ausschließlich für den Adressaten > bestimmt. Sollten Sie diese elektronische Nachricht irrtümlicherweise > erhalten haben, so informieren Sie uns bitte unverzüglich telefonisch oder > per E-Mail. > > This message is intended only for the use of the individual or entity to > which it is addressed. If you have received this message in error, please > delete the message and notify us immediately. > > > --------------070003040807050803020008 > Content-Type: text/html; charset=ISO-8859-1 > Content-Transfer-Encoding: 7bit > > <html> > <head> > <meta content="text/html; charset=ISO-8859-1" > http-equiv="Content-Type"> > </head> > <body bgcolor="#FFFFFF" text="#000000"> > Hi,<br> > <br> > you are missing libpcap0.8 which is a dependency for the captagent.<br> > <br> > <br> > BR<br> > <br> > Max M.<br> > <br> > Am 17.02.2012 16:31, schrieb Nelson Pereira: > <blockquote > cite=" > mid:CAC6cfSgZgOJQ0-X5ANsHEQnmt+AvV96404dU_nr53dTNdbcUoA@mail.gmail.com" > type="cite">ok, tried to do the make install, getting the bellow: > <div> > <div>aurast:/usr/local/src/captagent # > /usr/local/bin/captagent</div> > <div>/usr/local/bin/captagent: error while loading shared > libraries: libpcap.so.0.8: cannot open shared object file: No > such file or directory</div> > <div><br> > </div> > <div>Do i need to compile this?</div> > <div><br> > </div> > <div>Also just tried running the captagent:</div> > <div> > <div>aurast:/usr/local/src/captagent # ./captagent</div> > <div>./captagent: error while loading shared libraries: > libpcap.so.0.8: cannot open shared object file: No such file > or directory</div> > <div><br> > </div> > </div> > <div>What am I doing wrong?</div> > <div><br> > </div> > <div>This is a SUSE SLES11 64bit server running asterisk 1.8.6</div> > <div><br> > </div> > <div>Thanks</div> > <div><br> > </div> > <div><br> > </div> > <br> > <div class="gmail_quote"> > On Fri, Feb 17, 2012 at 10:24 AM, Nelson Pereira <span > dir="ltr"><<a moz-do-not-send="true" > href="mailto:kitkat0981@gmail.com">kitkat0981(a)gmail.com > </a>></span> > wrote:<br> > <blockquote class="gmail_quote" style="margin:0 0 0 > .8ex;border-left:1px #ccc solid;padding-left:1ex"> > from the atttached kamailio.cfg file, what should be changed > to get captagent receiver working in kamailio instead of > port mirroring? > <div><br> > </div> > <div>Thanks</div> > <div class="HOEnZb"> > <div class="h5"> > <div><br> > <br> > <div class="gmail_quote"> > On Fri, Feb 17, 2012 at 7:29 AM, Alexandr Dubovikov > <span dir="ltr"><<a moz-do-not-send="true" > href="mailto:voip@start4.info" target="_blank"> > voip(a)start4.info</a>></span> > wrote:<br> > <blockquote class="gmail_quote" style="margin:0 0 0 > .8ex;border-left:1px #ccc solid;padding-left:1ex"> > <div bgcolor="#FFFFFF" text="#000000"> <br> > The link has been not changed, you know it > already:<br> > <br> > <a moz-do-not-send="true" > href="http://code.google.com/p/homer/wiki/HOWTO" > target="_blank"> > http://code.google.com/p/homer/wiki/HOWTO</a><br> > <br> > Homer Capture Agent (for all other unsupported > system)<br> > <br> > if your SIP system (PBX, proxy) doesn't have a > built-in capture agent with HEP or IPIP > encapsulation support, you can use Homer Capture > Agent (captagent). This is an external > application (daemon) that collect SIP messages > thru PCAPlib and send them to your capture > server. The captagent support IPv4 and IPv6.<br> > <br> > preventing your next question: the captagent > your can find in the git.<br> > <br> > <a moz-do-not-send="true" > href=" > http://code.google.com/p/homer/source/browse/" > target="_blank"> > http://code.google.com/p/homer/source/browse/</a><br> > <br> > <br> > Wbr,<br> > Alexandr > <div> > <div><br> > <br> > <br> > 17.02.2012 13:15, Nelson Pereira wrote: > <blockquote type="cite"> > <div>Can captagent be installed on a > asterisk server? If so, is there a > howto?<br> > <br> > Sent from my iPhone4 on IOS4</div> > <div><br> > On Feb 17, 2012, at 7:07 AM, Alexandr > Dubovikov <<a moz-do-not-send="true" > href="mailto:voip@start4.info" > target="_blank">voip(a)start4.info</a>> > > wrote:<br> > <br> > </div> > <blockquote type="cite"> > <div> Hi,<br> > <br> > start two captagents instance, each on > own interface and duplicate all > traffic to your kamailio node. (HEP > protocol)<br> > <br> > Wbr,<br> > Alexandr<br> > <br> > 16.02.2012 17:51, Nelson Pereira > wrote: > <blockquote type="cite">I would like > to know if someone can show me an > example of there setup configuration > in kamailio.cfg for capturing > packets comming in from a mirror > port. > <div>The reason I ask is that if I > do a TCPDUMP on the server, I see > all traffic to and from different > sources on my Voice network.</div> > <div><br> > </div> > <div>But when I use Homer SIPCAPTURE > which uses Kamailio in the > background, I only see traffic > sent FROM my PBX but not seing the > traffic comming back to > it. </div> > <div><br> > </div> > <div>I mean I only see one side of > the call, yet TCPDUMP confirmed I > see everything....</div> > <div><br> > </div> > <div>What am I missing ?</div> > <div><br> > </div> > <div>Thanks</div> > <div><br> > </div> > <div>NP</div> > <br> > <fieldset></fieldset> > <br> > > <pre>_______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > <a moz-do-not-send="true" href="mailto:sr-users@lists.sip-router.org" > target="_blank">sr-users(a)lists.sip-router.org</a> > <a moz-do-not-send="true" href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" > target="_blank"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a> > </pre> > </blockquote> > <br> > </div> > </blockquote> > <blockquote type="cite"> > > <div><span>_______________________________________________</span><br> > <span>SIP Express Router (SER) and > Kamailio (OpenSER) - sr-users > mailing list</span><br> > <span><a moz-do-not-send="true" > href="mailto: > sr-users(a)lists.sip-router.org" > target="_blank"> > sr-users(a)lists.sip-router.org</a></span><br> > <span><a moz-do-not-send="true" > href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" > target="_blank"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > </a></span><br> > </div> > </blockquote> > <br> > <fieldset></fieldset> > <br> > > <pre>_______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > <a moz-do-not-send="true" href="mailto:sr-users@lists.sip-router.org" > target="_blank">sr-users(a)lists.sip-router.org</a> > <a moz-do-not-send="true" href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" > target="_blank"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a> > </pre> > </blockquote> > <br> > </div> > </div> > </div> > <br> > _______________________________________________<br> > SIP Express Router (SER) and Kamailio (OpenSER) - > sr-users mailing list<br> > <a moz-do-not-send="true" > href="mailto:sr-users@lists.sip-router.org" > target="_blank">sr-users(a)lists.sip-router.org > </a><br> > <a moz-do-not-send="true" > href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" > target="_blank"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br> > <br> > </blockquote> > </div> > <br> > </div> > </div> > </div> > </blockquote> > </div> > <br> > </div> > <br> > <fieldset class="mimeAttachmentHeader"></fieldset> > <br> > <pre wrap="">_______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > <a class="moz-txt-link-abbreviated" href="mailto: > sr-users(a)lists.sip-router.org">sr-users(a)lists.sip-router.org</a> > <a class="moz-txt-link-freetext" href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a> > </pre> > </blockquote> > <br> > <br> > <pre class="moz-signature" cols="72">-- > 42com Telecommunication GmbH > Straße der Pariser Kommune 12-16 / D-10243 Berlin > > E-mail: <a class="moz-txt-link-abbreviated" href="mailto:mm@42com.com"> > mm(a)42com.com</a> > Homepage: <a class="moz-txt-link-abbreviated" href="http://www.42com.com"> > www.42com.com</a> > > Firmenangaben/Company information: > > Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B > Umsatzsteuer-ID/VAT-ID: DE223812306, Geschäftsführer/CEO: Thomas > Reinig, Alexander Reinig > > > Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. > Diese sind möglicherweise vertraulich und ausschließlich > für den Adressaten bestimmt. Sollten Sie diese elektronische Nachricht > irrtümlicherweise erhalten haben, so informieren Sie uns bitte > unverzüglich telefonisch oder per E-Mail. > > This message is intended only for the use of the individual or entity to > which it is addressed. If you have received this message in error, please > delete the message and notify us immediately.</pre> > </body> > </html> > > --------------070003040807050803020008-- > > > _______________________________________________ > 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 1

Need example of Kamailio.cfg for a port mirror setup
by Nelson Pereira 17 Feb '12

17 Feb '12

I would like to know if someone can show me an example of there setup configuration in kamailio.cfg for capturing packets comming in from a mirror port. The reason I ask is that if I do a TCPDUMP on the server, I see all traffic to and from different sources on my Voice network. But when I use Homer SIPCAPTURE which uses Kamailio in the background, I only see traffic sent FROM my PBX but not seing the traffic comming back to it. I mean I only see one side of the call, yet TCPDUMP confirmed I see everything.... What am I missing ? Thanks NP

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Re: [SR-Users] how to match exact string value in mtree (Uri Shacked)
by Javier Gallart 17 Feb '12

17 Feb '12

Hello Uri I had similar needs and I found the ndb_redis module more suited for that type of task. Instead of a tree you have a hash like this: tname tprefix tvalue. If you do a hget nts $avp(DID) and you get a not null value you have found your exact match. It works very well for me and the time it takes for that "query" is barely noticeable. Of course redis does not address items like persistence, etc the same way a rdbms does. Hope it helps. Regards Javi On Thu, Feb 16, 2012 at 12:00 PM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. how to match exact string value in mtree (Uri Shacked) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 16 Feb 2012 12:54:50 +0200 > From: Uri Shacked <ushacked(a)gmail.com> > Subject: [SR-Users] how to match exact string value in mtree > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAMMbDhTFNXAE-K88=AeMjO7AnA_QJV3Ajj3AH-AxemXN3ze6HQ(a)mail.gmail.com > > > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > I am using Mtree to match prefix numbers, some of them starts with 0 or > characters like D for example. > so, the mtree param is like this: > > modparam("mtree", "db_url", CFGDB) > modparam("mtree", "mtree", "name=nts;dbtable=service_numbers_view;type=0;") > modpmodparam("mtree", "char_list", "0123456789*+#YMDabcdefgh") > modparam("mtree", "pv_value", "$avp(mtval)") > modparam("mtree", "pv_values", "$avp(mtvals)") > > The thing is, that i tried all: > > if(!mt_match("nts", "$avp(DID)","1")) > if(!mt_match("nts", "$avp(DID)","2")) > if(!mt_match("nts", "$avp(DID)","0")) > > and lets say i have the both prefix in the nts mtree: > 09555 > 09555333 > > And the prefix i search for is $avp(DID)=09555444 > > I allwasy get the 09555 because it is the longest match. > I need exact match..... > > how do i do that? >

3 7

ndb_redis module fails after a while
by Javier Gallart 17 Feb '12

17 Feb '12

Hi all I have started making some tests with the ndb_redis module. So far we have not stressed the module (no more than 5 HGET commands/second at maximum). It works well, but with at some point it starts failing. The failures are easily found because the logs always show this: INFO: <core> [main.c:811]: INFO: signal 13 received After that the redis value is always null. If I restart kamailio it starts working again. I've run kamailio with debug=4 but I haven't seen more useful information. On the redis side, I could find nothing in the logs either, the number of clientes connected is alway much less than the configured maximum, Any idea? On the other hand, if I restart redis we need to restart kamailio to restore the connections. Is the reconnection to redis on the roadmap? Thanks in advance Regards Javi

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Kamailio Exits With: Segmentation Fault Error 4 in usrloc.so
by Simpson Chua 17 Feb '12

17 Feb '12

Hi, Kamailio is exiting with a segmentation fault when trying to do a save("location"). Here is some information I gathered. Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: ERROR: <core> [db.c:435]: invalid parameter value Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: ERROR: usrloc [udomain.c:564]: failed to use table location Feb 16 15:30:19 ubuntu kernel: [171540.056663] kamailio[25721]: segfault at 89 ip 00007fa1f9a41452 sp 00007fff971a3af0 error 4 in usrloc.so[7fa1f9a37000+1f000] Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: ALERT: <core> [main.c:751]: child process 25720 exited by a signal 11 Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: ALERT: <core> [main.c:754]: core was generated Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: INFO: <core> [main.c:766]: INFO: terminating due to SIGCHLD Core was generated by `kamailio -w /home/core'. Program terminated with signal 11, Segmentation fault. #0 db_insert_ucontact (_c=0x7f5f8f737eb0) at ucontact.c:474 474 vals[11].val.str_val = _c->sock->sock_str; (gdb) bt #0 db_insert_ucontact (_c=0x7f5f8f737eb0) at ucontact.c:474 #1 0x00007f5f916d9649 in insert_ucontact (_r=<optimized out>, _contact=<optimized out>, _ci=<optimized out>, _c=0x7fff52bbeb88) at urecord.c:473 #2 0x00007f5f914b785f in insert_contacts (_m=0x7f5f93d7f458, _d=0x7f5f8f72f680, _a=0x7fff52bbec00) at save.c:428 #3 0x00007f5f914b8d10 in add_contacts (_mode=1, _a=0x7fff52bbec00, _d=0x7f5f8f72f680, _m=0x7f5f93d7f458) at save.c:737 #4 save (_m=0x7f5f93d7f458, _d=0x7f5f8f72f680, _cflags=4) at save.c:790 #5 0x000000000041ba87 in do_action (h=0x7fff52bbf5d0, a=0x7f5f93d7f238, msg=0x7f5f93d7f458) at action.c:1128 #6 0x000000000041abe2 in run_actions (h=0x7fff52bbf5d0, a=0x7f5f93d7f238, msg=0x7f5f93d7f458) at action.c:1610 #7 0x000000000041c05b in do_action (h=0x7fff52bbf5d0, a=0x7f5f93d7fe08, msg=0x7f5f93d7f458) at action.c:1107 #8 0x000000000041abe2 in run_actions (h=0x7fff52bbf5d0, a=0x7f5f93d7fe08, msg=0x7f5f93d7f458) at action.c:1610 #9 0x0000000000422fa2 in run_top_route (a=0x7f5f93d7fe08, msg=0x7f5f93d7f458, c=<optimized out>) at action.c:1683 #10 0x00000000004b061e in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:207 #11 0x00007f5f91b038b0 in raw_capture_rcv_loop (rsock=5, port1=5060, port2=8933, ipip=0) at sipcapture.c:1518 #12 0x00007f5f91b03e1f in init_rawsock_children () at sipcapture.c:572 #13 0x00007f5f91b04033 in child_init (rank=<optimized out>) at sipcapture.c:542 #14 0x0000000000507a0f in init_mod_child (m=0x7f5f93d7b2b0, rank=0) at sr_module.c:886 #15 0x0000000000507974 in init_mod_child (m=0x7f5f93d7b668, rank=0) at sr_module.c:883 #16 0x0000000000507974 in init_mod_child (m=0x7f5f93d7c178, rank=0) at sr_module.c:883 #17 0x0000000000507974 in init_mod_child (m=0x7f5f93d7c4b0, rank=0) at sr_module.c:883 #18 0x0000000000476149 in main_loop () at main.c:1661 #19 0x000000000041a940 in main (argc=<optimized out>, argv=0x7fff52bbfd68) at main.c:2475 Any idea why this is happening? Is there something in the REGISTER message that is causing this? Thanks, Simpson

2 2

Re: [SR-Users] how to match exact string value in mtree (Uri Shacked)
by Daniel-Constantin Mierla 17 Feb '12

17 Feb '12

Hello, On 2/17/12 1:04 PM, Javier Gallart wrote: > Hi Daniel, > > no, in order to find the best match we do a manual trick using > the s.prefixes transformation (I think you suggested that). could be, at least is the way I do prefix matching with mysql. Postgres, on the other hand, afaik, has an index plugin to match the longest prefix, might not be in default distro, but can be found on the net for sure. > We start with the full $rU and stop when we find a non-null value. > This kamailio is serving many thousands of calls per hour, and the > response time is pretty much the same as when we used the mtree > module. For us , using redis has the advantage of being able to load > the tree data faster than when we used postgres without the need to > perform a mtree reload. Loading huge amount of records from mysql takes time indeed. Maybe prefix matching will be added in redis in the future, or scripted server side in lua, as I understood it has such feature. Another option would be to add a custom command in kamailio like redis_cmd_pmatch() to do internally these prefix-based commands until a match, could simplify the config if it becomes something very popular. Also, I forgot about the other solution in kamailio - pdb module written by 1&1 for number portability: * http://kamailio.org/docs/modules/stable/modules/pdb.html Just mentioning it here for let it be known, if redis is working fine for you, makes no sense to look for other solutions. Cheers, Daniel > > Regards > > Javi > > > On Fri, Feb 17, 2012 at 10:25 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello Javier, > > one question about redis, can it do longest prefix matching or all > prefixes match? > > Cheers, > Daniel > > > On 2/16/12 2:51 PM, Javier Gallart wrote: >> Hello Uri >> >> I had similar needs and I found the ndb_redis module more suited >> for that type of task. Instead of a tree you have a hash like >> this: tname tprefix tvalue. If you do a hget nts $avp(DID) and >> you get a not null value you have found your exact match. It >> works very well for me and the time it takes for that "query" is >> barely noticeable. Of course redis does not address items like >> persistence, etc the same way a rdbms does. >> Hope it helps. >> >> Regards >> >> Javi >> >> On Thu, Feb 16, 2012 at 12:00 PM, >> <sr-users-request(a)lists.sip-router.org >> <mailto:sr-users-request@lists.sip-router.org>> wrote: >> >> Send sr-users mailing list submissions to >> sr-users(a)lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org> >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> or, via email, send a message with subject or body 'help' to >> sr-users-request(a)lists.sip-router.org >> <mailto:sr-users-request@lists.sip-router.org> >> >> You can reach the person managing the list at >> sr-users-owner(a)lists.sip-router.org >> <mailto:sr-users-owner@lists.sip-router.org> >> >> When replying, please edit your Subject line so it is more >> specific >> than "Re: Contents of sr-users digest..." >> >> >> Today's Topics: >> >> 1. how to match exact string value in mtree (Uri Shacked) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Thu, 16 Feb 2012 12:54:50 +0200 >> From: Uri Shacked <ushacked(a)gmail.com >> <mailto:ushacked@gmail.com>> >> Subject: [SR-Users] how to match exact string value in mtree >> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router >> (SER) - >> Users Mailing List" <sr-users(a)lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org>> >> Message-ID: >> <CAMMbDhTFNXAE-K88=AeMjO7AnA_QJV3Ajj3AH-AxemXN3ze6HQ(a)mail.gmail.com >> <mailto:AeMjO7AnA_QJV3Ajj3AH-AxemXN3ze6HQ@mail.gmail.com>> >> Content-Type: text/plain; charset="iso-8859-1" >> >> Hi, >> >> I am using Mtree to match prefix numbers, some of them starts >> with 0 or >> characters like D for example. >> so, the mtree param is like this: >> >> modparam("mtree", "db_url", CFGDB) >> modparam("mtree", "mtree", >> "name=nts;dbtable=service_numbers_view;type=0;") >> modpmodparam("mtree", "char_list", "0123456789*+#YMDabcdefgh") >> modparam("mtree", "pv_value", "$avp(mtval)") >> modparam("mtree", "pv_values", "$avp(mtvals)") >> >> The thing is, that i tried all: >> >> if(!mt_match("nts", "$avp(DID)","1")) >> if(!mt_match("nts", "$avp(DID)","2")) >> if(!mt_match("nts", "$avp(DID)","0")) >> >> and lets say i have the both prefix in the nts mtree: >> 09555 >> 09555333 >> >> And the prefix i search for is $avp(DID)=09555444 >> >> I allwasy get the 09555 because it is the longest match. >> I need exact match..... >> >> how do i do that? >>

2 1

get myself ip address on start
by Uri Shacked 17 Feb '12

17 Feb '12

Hi, i would like to get the kamailio server ip address on start up and save it in an $shv(). i tried to use "myself" but did manage to get the ip from it. any ideas? Uri

1 0

Re: [SR-Users] Kamailio Exits With: Segmentation Fault Error 4 in usrloc.so
by Daniel-Constantin Mierla 17 Feb '12

17 Feb '12

Hello, On 2/17/12 1:11 PM, Alexandr Dubovikov wrote: > Hi Daniel, > > he use raw_socket in the sipcapture module and save all REGISTER > messages to location. (something like fraud controling for > registration). I think in this case he have to ignore socket listener. not sure I understood -- it is on the server that saves to homer's database? It gets the traffic over a raw socket and then executes the config file as usual with functions from other modules? Cheers, Daniel > > Wbr, > Alexandr > > > > > 17.02.2012 09:19, Daniel-Constantin Mierla wrote: >> Hello, >> >> looks like an invalid listen socket structure, quite strange... what >> version of kamailio do you have? >> >> Also, in gdb, frame 0, can you send the output of: >> >> p *_c >> p *_c->sock >> >> You can replace sensitive data (like IP), if you wish. >> >> Cheers, >> Daniel >> >> On 2/16/12 10:52 PM, Simpson Chua wrote: >>> Hi, >>> >>> Kamailio is exiting with a segmentation fault when trying to do a >>> save("location"). Here is some information I gathered. >>> >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: DEBUG: >>> <core> [parser/msg_parser.c:103]: found end of header >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: ERROR: >>> <core> [db.c:435]: invalid parameter value >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: ERROR: >>> usrloc [udomain.c:564]: failed to use table location >>> Feb 16 15:30:19 ubuntu kernel: [171540.056663] kamailio[25721]: >>> segfault at 89 ip 00007fa1f9a41452 sp 00007fff971a3af0 error 4 in >>> usrloc.so[7fa1f9a37000+1f000] >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: ALERT: >>> <core> [main.c:751]: child process 25720 exited by a signal 11 >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: ALERT: >>> <core> [main.c:754]: core was generated >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: INFO: <core> >>> [main.c:766]: INFO: terminating due to SIGCHLD >>> >>> Core was generated by `kamailio -w /home/core'. >>> Program terminated with signal 11, Segmentation fault. >>> #0 db_insert_ucontact (_c=0x7f5f8f737eb0) at ucontact.c:474 >>> 474 vals[11].val.str_val = _c->sock->sock_str; >>> (gdb) bt >>> #0 db_insert_ucontact (_c=0x7f5f8f737eb0) at ucontact.c:474 >>> #1 0x00007f5f916d9649 in insert_ucontact (_r=<optimized out>, >>> _contact=<optimized out>, _ci=<optimized out>, _c=0x7fff52bbeb88) at >>> urecord.c:473 >>> #2 0x00007f5f914b785f in insert_contacts (_m=0x7f5f93d7f458, >>> _d=0x7f5f8f72f680, _a=0x7fff52bbec00) at save.c:428 >>> #3 0x00007f5f914b8d10 in add_contacts (_mode=1, _a=0x7fff52bbec00, >>> _d=0x7f5f8f72f680, _m=0x7f5f93d7f458) at save.c:737 >>> #4 save (_m=0x7f5f93d7f458, _d=0x7f5f8f72f680, _cflags=4) at save.c:790 >>> #5 0x000000000041ba87 in do_action (h=0x7fff52bbf5d0, >>> a=0x7f5f93d7f238, msg=0x7f5f93d7f458) at action.c:1128 >>> #6 0x000000000041abe2 in run_actions (h=0x7fff52bbf5d0, >>> a=0x7f5f93d7f238, msg=0x7f5f93d7f458) at action.c:1610 >>> #7 0x000000000041c05b in do_action (h=0x7fff52bbf5d0, >>> a=0x7f5f93d7fe08, msg=0x7f5f93d7f458) at action.c:1107 >>> #8 0x000000000041abe2 in run_actions (h=0x7fff52bbf5d0, >>> a=0x7f5f93d7fe08, msg=0x7f5f93d7f458) at action.c:1610 >>> #9 0x0000000000422fa2 in run_top_route (a=0x7f5f93d7fe08, >>> msg=0x7f5f93d7f458, c=<optimized out>) at action.c:1683 >>> #10 0x00000000004b061e in receive_msg (buf=<optimized out>, >>> len=<optimized out>, rcv_info=<optimized out>) at receive.c:207 >>> #11 0x00007f5f91b038b0 in raw_capture_rcv_loop (rsock=5, port1=5060, >>> port2=8933, ipip=0) at sipcapture.c:1518 >>> #12 0x00007f5f91b03e1f in init_rawsock_children () at sipcapture.c:572 >>> #13 0x00007f5f91b04033 in child_init (rank=<optimized out>) at >>> sipcapture.c:542 >>> #14 0x0000000000507a0f in init_mod_child (m=0x7f5f93d7b2b0, rank=0) >>> at sr_module.c:886 >>> #15 0x0000000000507974 in init_mod_child (m=0x7f5f93d7b668, rank=0) >>> at sr_module.c:883 >>> #16 0x0000000000507974 in init_mod_child (m=0x7f5f93d7c178, rank=0) >>> at sr_module.c:883 >>> #17 0x0000000000507974 in init_mod_child (m=0x7f5f93d7c4b0, rank=0) >>> at sr_module.c:883 >>> #18 0x0000000000476149 in main_loop () at main.c:1661 >>> #19 0x000000000041a940 in main (argc=<optimized out>, >>> argv=0x7fff52bbfd68) at main.c:2475 >>> >>> Any idea why this is happening? Is there something in the REGISTER >>> message that is causing this? >>> >>> Thanks, >>> Simpson >>> >>> >>> _______________________________________________ >>> 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://linkedin.com/in/miconda -- http://twitter.com/miconda >> >> >> _______________________________________________ >> 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://linkedin.com/in/miconda -- http://twitter.com/miconda

2 3

Question on ("sipcapture", "raw_interface", "")
by Nelson Pereira 17 Feb '12

17 Feb '12

Is it possible to monitor 2 interfaces? I have 2 Asterisk servers and they are on 2 different switches so I need to capture ETH0 and ETH3 which is where the mirror ports are connected to from each switch. Thanks

3 3

CAcert root in Lumicall
by Daniel Pocock 17 Feb '12

17 Feb '12

I've been contemplating Daniel's earlier question about using the CAcert certificates with Lumicall sip5060.net should already accept mutual authentication from other Kamailio instances running with a CAcert certificate However, the Lumicall dialer itself will only connect to servers that are using a cert signed by a root CA trusted within Android. This applies to both the SIP and STUN over TLS support. It may be possible for me to embed the CACert root in the Lumicall package and create a custom trust manager - allowing Lumicall to work with any server using CAcert TLS certificates. Does anyone have any strong feelings about not doing something like that?

3 9

how to match exact string value in mtree
by Uri Shacked 17 Feb '12

17 Feb '12

Hi, I am using Mtree to match prefix numbers, some of them starts with 0 or characters like D for example. so, the mtree param is like this: modparam("mtree", "db_url", CFGDB) modparam("mtree", "mtree", "name=nts;dbtable=service_numbers_view;type=0;") modpmodparam("mtree", "char_list", "0123456789*+#YMDabcdefgh") modparam("mtree", "pv_value", "$avp(mtval)") modparam("mtree", "pv_values", "$avp(mtvals)") The thing is, that i tried all: if(!mt_match("nts", "$avp(DID)","1")) if(!mt_match("nts", "$avp(DID)","2")) if(!mt_match("nts", "$avp(DID)","0")) and lets say i have the both prefix in the nts mtree: 09555 09555333 And the prefix i search for is $avp(DID)=09555444 I allwasy get the 09555 because it is the longest match. I need exact match..... how do i do that?

2 1

image missing in xmlrpc module doc
by Andrew Pogrebennyk 17 Feb '12

17 Feb '12

Hi, Was reading through the xmlrpc docs, could anybody with an access check why Figure "RPC Example" is missing from http://kamailio.org/docs/modules/devel/modules/xmlrpc.html#fig.rpc_example as well as 3.1 and 3.2 module docs? Thanks, Andrew

1 0

Kamailio in VUC session, next Friday, Feb 17
by Daniel-Constantin Mierla 17 Feb '12

17 Feb '12

Hello, there will be a VoIP Users Conference (VUC) session dedicated to Kamailio SIP Server next week, Friday, February 17, starting with 17:00GMT. VUC is an online weekly audio conferencing event, you can connect via SIP and IRC to interact with the other participants. More details (including the ways to connect) are available at: * http://www.kamailio.org/w/2012/02/kamailio-at-voipuserconference-fri-feb-17… Join the event to learn about latest developments and future plans for the project, ask questions about your interested topics. Spreading the word about the event is also appreciated. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 2

mi_xmlrpc not compiling on Solaris
by Nathaniel L Keeling 17 Feb '12

17 Feb '12

Hello, I am trying to compile mi_xmlrpc on Solaris 10 64bit OS and I am getting this error: CC (gcc) [M mi_xmlrpc.so] abyss_xmlrpc_server.o abyss_xmlrpc_server.c:948:1: error: conflicting types for 'shutdownAbyss' abyss_xmlrpc_server.c:945:34: note: previous declaration of 'shutdownAbyss' was here abyss_xmlrpc_server.c: In function 'normalLevelAbyssRun': abyss_xmlrpc_server.c:1005:42: warning: passing argument 2 of 'xmlrpc_registry_set_shutdown' from incompatible pointer type [enabled by default] /opt/csw/include/xmlrpc-c/server.h:114:1: note: expected 'void (* const)(struct xmlrpc_env * const, void * const, const char * const, void * const)' but argument is of type 'void (*)(struct xmlrpc_env * const, void * const, const char * const)' gmake[1]: *** [abyss_xmlrpc_server.o] Error 1 gmake: *** [modules_k] Error 1 Thanks Nathaniel L Keeling

3 2

How to do request URI rewrites using database tables?
by Ben WIlliams 16 Feb '12

16 Feb '12

Hi, can someone please recommend the most appropriate modules to rewrite R-URIs based on a database lookup table? I've read the documentation for lcr and carrierroute but not sure if they can do this. In most cases it will be a simple R-URI rewrite but I also need to rewrite based on From user. ie R-URI match From match new R-URI =========== ========== ========= *97(a)example.com 6[0-9](a)example.com *97(a)10.0.0.1 *97(a)example.com 7[0-9](a)example.com *97(a)10.0.0.2 6[0-9](a)example.com match any rewrite domain to 10.0.0.1 abc(a)example.com match any def(a)example.com Thanks Ben

1 1

Problems with TCP
by Mino Haluz 16 Feb '12

16 Feb '12

Hi, our customers are using mostly UDP but some of them want to use TCP. The problem is, I get various TCP errors in kamailio log and I do not understand what they mean 1) ERROR: <core> [tcp_main.c:4130]: connect <customerIP:port> failed This means kamailio is trying to connect with TCP to the customer ?? 2) WARNING: tcp_do_connect <customerIP:port> : could not find corresponding listening socket for IPaddress , using default... ( where IPaddress is kamailio physical IP address) Kamailio has one virtual IP address (heartbeat IP address) and one physical. It listens on the virtual IP address but I do not understand why it is trying to use the physical IP address ? I was investigating this thing because the BYE message is not being relayed if the TCP is used (everytime the 2. error message is printed), so that's why I have to understand what it really means... And the last question, 3) If kamailio is relaying the traffic to the PSTN gateway, and the customer is using TCP, kamailio uses TCP as well for connecting to the PSTN gw. Is this behaviour normal ? I would expect that customer -- TCP --> kamailio --- UDP --> GW, and not customer -- TCP --> kamailio --- TCP --> GW, Thanks, Mino

3 3

Kamailio Asterisk Followup
by Greg Mannie 16 Feb '12

16 Feb '12

Hello Sorry for another newbie question, but eventually with your "greatly appreciated" help I will get proficient in this application. After reading much RFC reading and docs for Kamailio I see where the benefits of using the sip proxy for registering devices while using asterisk for voicemail or ivr etc. has great benefit. I am not finding much on my end user interaction. If I use realtime integration and have multi domain use on kamailio, how do I allow the end user to configure their own IVR? Is it possible to use modules like the call flow control from freepbx and allow users to configure this themselves? Regards, Greg

4 4

Current Vacancy
by apache＠iptel.org 16 Feb '12

16 Feb '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Job Category: Customer Support/Client Care Reference Code: HJR/EU/551-188 Job Status: Full Time, Temporary/Contract/Project Great company represented in 32 countries worldwide is seeking a Customer Service Representative. This is a temporary position with the great opportunity of going permanent. The responsibilities are as follows: assist to work with existing clients, arrange and assure preparation, final inspection, timely implementation of orders as well as all related documentation and paperwork. [...] Content analysis details: (20.7 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 1.3 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net [Blocked - see <http://www.spamcop.net/bl.shtml?89.252.8.22>] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [89.252.8.22 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 0.8 RCVD_IN_SORBS_WEB RBL: SORBS: sender is an abusable web server [89.252.8.22 listed in dnsbl.sorbs.net] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [89.252.8.22 listed in bb.barracudacentral.org] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: eurojoboard.com] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: eurojoboard.com] 3.6 HELO_DYNAMIC_IPADDR2 Relay HELO'd using suspicious hostname (IP addr 2) 3.5 HELO_DYNAMIC_SPLIT_IP Relay HELO'd using suspicious hostname (Split IP) 0.0 TVD_RCVD_IP TVD_RCVD_IP 1.3 DATE_IN_PAST_24_48 Date: is 24 to 48 hours before Received: date 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.5553] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

strage result in math with $vars in 3.2.2
by Uri Shacked 16 Feb '12

16 Feb '12

Hi, i am using some math to send to my NOC the memory usage of kamailio. this is what i do: $var(xxx)=$stat(real_used_size)*100/$stat(total_size) before the 3.2.2 upgrade, i usualy got regular results - 3,5,20 etc.... now, i get results like -8, -32 and so... any ideas why? here are the $stat numbers that gets me the -8 result: $stat(real_used_size)=31899516 $stat(total_size)=134217728

2 3

How to reload dispatcher destinations
by Mino Haluz 16 Feb '12

16 Feb '12

Hi, is there any way how to reload dispatcher destinations (located in db) without restarting the kamailio when algorithm 10 (call load balancing) is used ? I read in documentation, that this command is disabled when this algorithm is used. Thanks, Mino

2 2

some questions
by 393319152 16 Feb '12

16 Feb '12

dear friends: I'm using the kamailio as my sip server and it has a public ip. There are two nac: A and B, A has a privite ip, B has a privite ip that is seted with VPN . And now, A can register successfully, but B can't . I want A to connect B and trans information. I don't know why , I hope you can help me.

2 1

Kamailio v3.2.2 Released
by Daniel-Constantin Mierla 15 Feb '12

15 Feb '12

Hello, Kamailio SIP Server v3.2.2 stable release is out. This is a maintenance release of the latest stable branch, 3.2, that includes fixes since release of v3.2.1. There is no change to database schema or configuration language structure that you have to do on installations of v3.2.0 or v3.2.1. Deployments running previous v3.2.x versions are strongly recommended to be upgraded to v3.2.2. For more details about version 3.2.2 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/01/kamailio-v3-2-2-released/ RPM packages are currently on the build service, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 3

Reuse TLS connections
by Bruno Bresciani 15 Feb '12

15 Feb '12

Hi all, There is the possibility to kamailio reuse the TLS connection created by other SIP server? When kamailio use t_relay function to send a SIP request message to other server, kamailio verify if already exist some connection established with the destiny and use it even if this connection was created by the other SIP server. In short, I want to keep only one connection between kamailio and SIP server. Sometimes kamailio will be a client and others a server. Best Regards

2 3

struct cmd_export_t
by Bruno Bresciani 15 Feb '12

15 Feb '12

Hi all, I'm writing a module to kamailio and can't find detail information about the parameter *"fixup_function fixup"* present at struct *cmd_export_t*. I can't understand which functionality this parameter... Someone have any information or documentation about this subject. Best Regards

2 2

AVP obtain with regular expression
by Anto 14 Feb '12

14 Feb '12

Hello I'm using AVP, but I can not finish assimilating the concept. I try to input messages, according to their DID. Once it enters an INVITE message treatment with alias_db for its user, but I want some AVP for each DID. I can think of three cases. The first I have no very clear if possible and if it were, I think is not optimal. 1) Given the configuration of the module: modparam("avpops","avp_url","mysql://user:passwd@host/database") Had thought if I can use regular expressions to get the "prs" in the database (DID_IN, DID_DEST, etc). USR_PREFERENCES table: | id | uuid | username | domain | attribute | type | value | modified | | 1 | XXX | user1 | dmn.tld | 571XXX_IN | 1 | XXX | | | 2 | XXX | user2 | dmn.tld | 571XXX_DEST | 1 | XXX | | 2) Maybe better to use the module configuration schemes: modparam("avpops","db_scheme","scheme1:uuid_col=uid;value_col=job;value_type=string;table=emp") modparam("avpops","db_scheme","scheme2:username_col=user;domain_col=domain;value_col=email;table=users") 3) Or use avp_db_query() function. Can anyone guide me? I'm pretty lost :-S. The first case, I know if possible and possibly more difficult to maintain. Thanks very much. Best regards Anto

1 0

problems since upgrading mysql and php
by Nelson Pereira 14 Feb '12

14 Feb '12

Getting the bellow error after upgrading php and mysql as requested: Access forbidden! You don't have permission to access the requested directory. There is either no index document or the directory is read-protected. If you think this is a server error, please contact the webmaster <mailto:root@cbnhmr01plc> . Error 403 172.20.25.149 <http://172.20.25.149/> Tue Feb 14 13:37:07 2012 Apache/2.2.10 (Linux/SUSE) Regards, Nelson Pereira Sr. Network Administrator, Information Systems Canadian Bank Note Company 18 Auriga Drive Ottawa, Ontario K2E 7T9 Phone: (613) 225-6566 Ext 2347 Cell: (613) 410-7608

2 1

Record-route and high avaibility
by Anto 14 Feb '12

14 Feb '12

Hello I have a proxy that makes the function of balancer and I have a doubt. This balancing between two proxies that do accounting, if any falls during the course of a call, you lose accounting (record-route). Is that correct? Is it advisable to have at each proxy server as a slave and balance the ip? Thanks Regards

2 2

creating DB in MySQL for Homer Capture server gives errors
by Nelson Pereira 14 Feb '12

14 Feb '12

Hi all, Trying to create the Homer SIP capture (Kamailio) DB and getting an error: Error SQL query: CREATE TABLE `sip_capture` ( `id` bigint(20) unsigned NOT NULL AUTO_INCREMENT, `date` timestamp NOT NULL DEFAULT '0000-00-00 00:00:00', `micro_ts` bigint(18) NOT NULL DEFAULT '0', `method` varchar(50) NOT NULL DEFAULT '', `reply_reason` varchar(100) NOT NULL, `ruri` varchar(200) NOT NULL DEFAULT '', `ruri_user` varchar(100) NOT NULL DEFAULT '', `from_user` varchar(100) NOT NULL DEFAULT '', `from_tag` varchar(64) NOT NULL DEFAULT '', `to_user` varchar(100) NOT NULL DEFAULT '', `to_tag` varchar(64) NOT NULL, `pid_user` varchar(100) NOT NULL DEFAULT '', `contact_user` varchar(120) NOT NULL, `auth_user` varchar(120) NOT NULL, `callid` varchar(100) NOT NULL DEFAULT '', `callid_aleg` varchar(100) NOT NULL DEFAULT '', `via_1` varchar(256) NOT NULL, `via_1_branch` varchar(80) NOT NULL, `cseq` varchar(25) NOT NULL, `diversion` varchar(256) NOT NULL, `reason` varchar(200) NOT NULL, `content_type` varchar(256) NOT NULL, `authorization` varchar(256) N[...] MySQL said: <http://172.20.25.149/phpMyAdmin/url.php?url=http%3A%2F%2Fdev.mysql.com%2Fdo c%2Frefman%2F5.0%2Fen%2Ferror-messages-server.html&token=1d6b9d77bce4c6f1648 83e3c6dd6bc09> Description: Documentation #1064 - You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'PARTITION BY RANGE ( UNIX_TIMESTAMP(`date`) ) ( PARTITION pmax VALUES LESS THAN ' at line 50 System Information: . Apache/2.2.10 (Linux/SUSE) . MySQL client version: 5.0.67 . PHP extension: mysqli Server is SLES 11 SP1

2 1

get server parameters from DB when kamailio starts
by Uri Shacked 14 Feb '12

14 Feb '12

Hi again... (i do find myself ask many questions these days.... :-)) well, now here is one more: i have 2 kamailio servers that i use in 2 locations. both use the same tables and DBURL. each of my server turns to a number portability (NP) server at its same location. for example, if server 1 has the call, it will turn to NP server 1 with the query, and if the query fails, it will then turn to NP server 2. the kamailio server 2 does the same. but, turns to NP server 2 first and if it fails, to NP server 1. once this is understood.... here is my issue - i want to have 1 generic cfg file for my 2 servers. today, i use the same cfg file, but in server 1 i set it in the cfg to its group, and on server 2 the same but a different group.... i tried to understand how do i get each server to "pull" its group from the DB. something like "what is my group when my IP is X.Y.Z.Q ? i raed the cfg_db module but didn't understand how to set it on my cfg file and on the DB. what is the best way to do what i want? hope i was clear.... BR, thanks to all the guys that help me so far, i appriciate it a lot! Uri

3 8

London Meetup in March, at UC Expo
by Daniel-Constantin Mierla 14 Feb '12

14 Feb '12

Hello, it is very likely I will be in London and visit UC Expo show (http://www.ucexpo.co.uk/) at least the first day. Therefore I am thinking of organizing a Kamailio/SIP/VoIP social networking event (perhaps a dinner or pub meeting), on the evening of the 5th or 6th of March. Are people from community interested in joining such event? If yes, write me your preferred date -- based on feedback in the next days, it will be decided whether it is going to happen and when. Also, if many people would be interested in a project developments and updates session (like the one did in 2010: http://www.kamailio.org/w/present-and-future-of-sip-routing-2010-london/), I can try to get a location for 2-3 hours on the 5th of March for such meetup. Let me know as well your interest in such event. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

get myself ip address on start
by Uri Shacked 14 Feb '12

14 Feb '12

Hi, i would like to get the kamailio server ip address on start up and save it in an $shv(). i tried to use "myself" but did manage to get the ip from it. any ideas? Uri

1 0

Is there something like "silence detect hangup"
by Stoyan Mihaylov 13 Feb '12

13 Feb '12

Problem - if connection drop, call can persist. In Asterisk there is silencedetecthangup - which should cause hangup, if there is full silence for desired period of time. Unfortunately it does not hangup. I mean: SIP client 1 ->Kamailio -> Asterisk ->Kamailio -> SIP client 2 If I drop connection for one of SIP clients, I expect call should be automatically hangup after a time I set (20 sec). But call persists. In worst case - if connection drops for both clients, call will persist until Asterisk is restarted. I will continue to look how to solve problem with Asterisk, but I am curious if this can be done from Kamailio also. If I can cancel call from both places - it will be great. I need to ensure that if something wrong happens, call will be dropped within 30 sec maximum. Stoyan

2 2

Is exec module ignoring time_to_kill parameter?
by Miguel Baptista 13 Feb '12

13 Feb '12

Hello, I am running kamailio 3.1.3 and I am making some tests with the exec module. I am running a "dummy" script and it worked like a charm. Now, I am testing what happens when the script takes "too much time" to return a result. According to exec module documentation the/ time_to_kill /parameter should take care of it. *time_to_kill* (INTEGER) /Specifies the longest time a program is allowed to execute. If the time is exceeded, the program is killed. / BTW ... I am assuming /time /is in milliseconds. I think the documentation should refer the time units. Apparently kamailio ignores this parameter (or it doesn't work as I am expecting). No matter which value I assign to /time_to_kill/, kamailio doesn't kill the script and waits for the script to finish. In the meanwhile some SIP timers are triggered ... resulting into a failed call. The test I am doing is quite simple. For incoming INVITEs, use a external script to check the status of a specific test user, if it returns the keyword "busy" sends it to voicemail otherwise route the INVITE to the local user. The test script returns a result after about 8 seconds. Shouldn't the /time_to_kill/ be triggered and stop the script? Here goes some relevant parts of my configuration. In attachment I am sending the kamailio log with a debug value of 4. /# ----- exec params -----/ /modparam("exec", "time_to_kill", 10)/ /# --- exec params end -----/ /#route just to check the exec module route[TESTEXEC] { xlog("L_INFO", "TESTEXEC: Entered ($rU)\n"); if ($rU == "miguel") { $var(calendarTMP)="/root/" + $rU + "_test.pl"; exec_avp("$var(calendarTMP)", "$avp(s:test)"); xlog("L_INFO", "TESTEXEC: Returned value is $avp(s:test)\n"); if ($avp(s:test) == "busy") { $ru = "sip:userbusy@" + ASTSERVIPPORT; # Voicemail route(RELAY); } } xlog("L_INFO", "TESTEXEC: Exiting ($rU)\n"); return; } / Regards, Miguel Baptista

2 2

TCP keepalive on server and client
by Yufei Tao 13 Feb '12

13 Feb '12

Hi I want to use TCP keepalive to detect dead connections between Kamailio and my SIP clients. I've got some questions about this: 1. If I enable it on Kamailio by setting for example: tcp_crlf_ping = yes tcp_keepalive = yes tcp_keepidle = 60 tcp_keepintvl = 10 tcp_keepcnt = 3 >From what I read by gooling, that means Kamailio only responds to the crlfcrlf sent by the client, but can't be configured to send the crlfcrlf. Is this right? 2. On my client, if I use setsockopt to set the corresponding values: SO_KEEPALIVE TCP_KEEPCNT TCP_KEEPIDLE TCP_KEEPINTVL would it make the client send crlfcrlf? The reason I'm confused is, with these settings, if Kamailio can only respond to crlfcrlf, but not send, then would the client only respond too, as in 1. and 2., they seem to configure the same set of parameters? Please help me clarify this. 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

3 2

Kamailio to Skype/Gtalk/Yahoo
by Abid Naseem 13 Feb '12

13 Feb '12

Hi, Just wanted to know if it is possible for kamailio to route calls to user's skype/gtalk/yahoo messenger? -- Regards, *Abid Naseem* +44 (793) 741 5043 +1 (415) 562 6333

2 1

Howto configure to use SQLITE
by Perazim 13 Feb '12

13 Feb '12

I am a new user and am trying to configure kamailio to use SQLITE instead of MYSQL. I see that SQLITE option is rather new and maybe not as documented as MYSQL. I am running on fedora 14 and have installed MYSQL3 from rpms. Kamailio I built from sources because I need the SQLITE support. What kamailio.cfg and kamctlrc parameters do I need to change to do this? I have already set these in kamctlrc: ... ## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT, or SQLITE # by default none is loaded # # If you want to setup a database with kamdbctl, you must at least specify # this parameter DBENGINE=SQLITE ## database host DBHOST=localhost ## database name (for ORACLE this is TNS name) # DBNAME=openser ... and ran: kamdbctl create openser and kamctl add user1 password1 successfully. Now, it appears that I need to change kamailio.cfg for SQLITE. Can I simply edit this file substituting SQLITE for MYSQL and loading the SQLITE module? How about this part: ... ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif ... Will this URL work with SQLITE? Thanks in advance, Perazim

3 3

OT: Sipwise raises Series A round for Kamailio-based soft-switch
by Andreas Granig 11 Feb '12

11 Feb '12

Hey guys, We're thrilled to announce that we closed our Series A financing round for our Kamailio-based soft-switch: http://www.sipwise.com/news/sipwise_series_a/ We're desperately looking for Voip engineers with good skills in SIP, Kamailio and/or Sems, so if you're interested in improving our open-source solution based on Kamailio, Sems, Asterisk etc, please contact me! Best regards, Andreas

1 0

Detection of broken TCP connections to clients
by Yufei Tao 10 Feb '12

10 Feb '12

Hi When clients register to Kamailio over TCP/TLS, if I set tcp_connection_lifetime to be quite small, like 30 seconds, and let clients send refreshes every 20 seconds, for example. When client crashes or have network problems, the TCP connection with Kamailio will be closed by the server after 30 seconds of being idle. My question is, is there a way of letting Kamailio tell which user the broken TCP connection belongs to, and then remove the user from the location table and update presence information etc? The reason I want this is, if a user publishes its status of being online with expiring time of 1 hour, but then crashed, then the location table entry and presence information of this user would be wrong for the rest of the hour. How do I get around this? I assume server can send periodic OPTION messages but that would be a bit expensive. Thank you 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

3 3

Re: [SR-Users] dialog vars issue
by Daniel-Constantin Mierla 09 Feb '12

09 Feb '12

Hello, back on mailing list ... I think I got it -- the acc record is written when 200 ok for invite is sent out, dialog is processing it before. Therefore dialog structure is updated with to-tag from the reply, but the building of acc record is executed using the INVITE request taken from transaction, where is no to-tag (acc internal design). This results in failing getting the dialog, because of to-tag mismatch. I will send a patch soon. Cheers, Daniel On 2/8/12 2:39 PM, Jon Bergli Heier wrote: > Hi, this is a gzipped log from a test call using kamailio 3.2.1. There's some noise in the log from REGISTERs and so on. > The dlg_var I'm setting is pzzid, printed as zid=<value>. > > Thanks, > Jon -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 2

Howto configure to use SQLITE
by Perazim 09 Feb '12

09 Feb '12

I am a new user and am trying to configure kamailio to use SQLITE instead of MYSQL. I see that SQLITE option is rather new and maybe not as documented as MYSQL. I am running on fedora 14 and have installed MYSQL3 from rpms. Kamailio I built from sources because I need the SQLITE support. What kamailio.cfg and kamctlrc parameters do I need to change to do this? I have already set these in kamctlrc: ... ## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT, or SQLITE # by default none is loaded # # If you want to setup a database with kamdbctl, you must at least specify # this parameter DBENGINE=SQLITE ## database host DBHOST=localhost ## database name (for ORACLE this is TNS name) # DBNAME=openser ... and ran: kamdbctl create openser and kamctl add user1 password1 successfully. Now, it appears that I need to change kamailio.cfg for SQLITE. Can I simply edit this file substituting SQLITE for MYSQL and loading the SQLITE module? How about this part: ... ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif ... Will this URL work with SQLITE? Thanks in advance, Perazim

1 0

Kamailio Socket
by Reda Aouad 09 Feb '12

09 Feb '12

Is this normal on restarting Kamailio ? ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) RA

2 1

get server parameters from DB when kamailio starts
by Uri Shacked 08 Feb '12

08 Feb '12

Hi again... (i do find myself ask many questions these days.... :-)) well, now here is one more: i have 2 kamailio servers that i use in 2 locations. both use the same tables and DBURL. each of my server turns to a number portability (NP) server at its same location. for example, if server 1 has the call, it will turn to NP server 1 with the query, and if the query fails, it will then turn to NP server 2. the kamailio server 2 does the same. but, turns to NP server 2 first and if it fails, to NP server 1. once this is understood.... here is my issue - i want to have 1 generic cfg file for my 2 servers. today, i use the same cfg file, but in server 1 i set it in the cfg to its group, and on server 2 the same but a different group.... i tried to understand how do i get each server to "pull" its group from the DB. something like "what is my group when my IP is X.Y.Z.Q ? i raed the cfg_db module but didn't understand how to set it on my cfg file and on the DB. what is the best way to do what i want? hope i was clear.... BR, thanks to all the guys that help me so far, i appriciate it a lot! Uri

1 0

finally, an open alternative to Viber
by Daniel - Lumicall.org 08 Feb '12

08 Feb '12

The rise of Viber has, for some people, been a case of `skype, not again!' until now... Lumicall is now in the Android market - and it fully interacts with other SIP products using ENUM and SRV records. Any feedback on this is welcome. Interconnect (Kamailio calling to/from Lumicall users) is based on the idea of federated SIP, it is explained at http://www.lumicall.org for those who want to connect up to it. Please bear in mind: Lumicall supports ICE (RFC 5245) for NAT traversal, it is using the ice4j implementation from the Jitsi community. This makes the SIP packets bigger and often they are too big for the MTU of a UDP packet. When using ICE, it seems essential to use TLS, to avoid the MTU problems and also to avoid routers mangling the SIP headers (ICE doesn't need help from routers, they only confuse the algorithm)

2 2

What is the valid values for flags field in dispatcher table?
by Gary Chen 08 Feb '12

08 Feb '12

Trying to figure out the values for flags field. where can I find the meaning of flags for each value like: flags 1 = ? flags 2 = ? flags 3 = ? flags 4 = ? flags 5 = ? flags 6 = ? flags 7 = ? flags 8 = ? Gary

2 1

Re: [SR-Users] "kamctl cr show" when carrierroute table has another name
by Uri Shacked 08 Feb '12

08 Feb '12

OK i got it. on the kamctl, the show is a SQL and the reload is $CTLCMD this is the difference... thanks On Wed, Feb 8, 2012 at 7:45 PM, Henning Westerholt <hw(a)kamailio.org> wrote: > On Wednesday 08 February 2012, Uri Shacked wrote: > > Can u explain why does the reload takes the table name from cfg and > > the show from the kamctlrc ? > > Hi Uri, > > the reload is executed in the server, which already uses the correct table. > The show is executed from the script (its just a database operation, you > could > also use a mysql client), which uses the wrong table. > > Viele Grüße/ best regards, > > Henning Westerholt >

1 0

Re: [SR-Users] "kamctl cr show" when carrierroute table has another name
by Uri Shacked 08 Feb '12

08 Feb '12

> > > anyone for this? > > Hi, > > after resolving the field type issue :-), i changed the params in kamctlrc > to point to the right DB. > now, i notice that when i use the command "kamctl cr show" iget an error > that it does not find the table "carrierroute". > in the cfg file it is all set and works fine, cr dump is working as > expected as well. > > in the kamctl file i can find the param - CARRIERROUTE_TABLE. but where is > it set from? > anyone knows where do i make tha change in order for the kamctl to "know" > the names of the tables i use? > > thanks, > Uri >

2 3

(no subject)
by Jon Bergli Heier 08 Feb '12

08 Feb '12

Hi, since kamailio 3.2.1 I've had problems with using dlg_var to store extra accounting fields with the acc module (using db_extra and mysql). When I set a dlg_var in the request route, the first acc row (INVITE) appears as if the dlg_var was empty, but looks fine for the second (BYE) row. In 3.2.0 both rows got the value I assigned. missed_calls doesn't appear to have this problem. Any ideas? The problem also seems to exist in kamailio 3.2.2. I've done a git bisect and the problem seems to appear since commit 5bdee1ffe820d3e9e75898bb252d1660b4bbad60. Thanks, Jon

2 3

best way to start kamailio at boot
by Uri Shacked 08 Feb '12

08 Feb '12

Hi, any help will be graet.... * hi, *>* *>* when i moved the /etc/init.d/kamailio file. kamailio did not start at boot. *>* now, i set the debug level to 4 and run kamailio in both ways. *>* *>* 1. "/etc/init.d/kamailio start" *>* * the resault was : *Feb 1 19:59:04 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:202]: qm_malloc_init: QM_OPTIMIZE=16384, /ROUNDTO=2048 Feb 1 19:59:04 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:204]: qm_malloc_init: QM_HASH_SIZE=2099, qm_block size=218344 Feb 1 19:59:04 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:205]: qm_malloc_init(0x7f1cf585b000, 2147483648), start=0x7f1cf585b000 Feb 1 19:59:04 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:213]: qm_malloc_init: size= 2147483648, init_overhead=218440 Feb 1 19:59:04 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/shm_mem.c:203]: shm_mem_init: success* >* *>* you can see the shm_mem size is 2048Mb. *>* * 2."kamailio" the resault was : *Feb 1 19:56:50 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:202]: qm_malloc_init: QM_OPTIMIZE=16384, /ROUNDTO=2048 Feb 1 19:56:50 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:204]: qm_malloc_init: QM_HASH_SIZE=2099, qm_block size=218344 Feb 1 19:56:50 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:205]: qm_malloc_init(0x7f8618310000, 33554432), start=0x7f8618310000 Feb 1 19:56:50 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:213]: qm_malloc_init: size= 33554432, init_overhead=218440 Feb 1 19:56:50 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/shm_mem.c:203]: shm_mem_init: success* ** the resault is 32Mb. any ideas? >* *>* On Wed, Feb 1, 2012 at 5:01 PM, Jason Penton <jason.penton at smilecoms.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >wrote: *>* *>>* I am not familiar with Centos but I suspect there is another script being *>>* run from somewhere. What I would do is move the /etc/init.d/kamailio script *>>* somewhere safe (your home directory for eg). Then reboot and see if *>>* kamailio still starts - if it does you need to find out whihc script is *>>* being called and where it is. *>>* *>>* If you are not running X by default you are prob in runlevel 3. If you *>>* are running X, then it will be 5. check where you runlevel scripts are *>>* pointing. e.g. *>>* *>>* ls -al /etc/rc3.d/SXXkamailio (X = [0-9]) *>>* *>>* Cheers *>>* Jason *>>* *>>* On Wed, Feb 1, 2012 at 4:26 PM, Uri Shacked <uri.shacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >wrote: *>>* *>>>* Hi, *>>>* *>>>* when i reboot the machine, it starts with 32Mb and i do not understand *>>>* why... *>>>* so, does it use a startup file that i am not familiar with? *>>>* what is the appropriate runlevel? how do i set it and where? *>>>* *>>>* thanks, *>>>* Uri *>>>* *>>>* *>>>* >Hi Sammy, *>>>* > *>>>* >When you use the init script, it is starting Kamailio with the correct *>>>* >arguments to get the environment you wish (-m 2048 probably). If *>>>* however, *>>>* >you just call kamailio these args are not there. Im not sure why you are *>>>* >running with kamailio command directly. Why don't you add your init *>>>* script *>>>* >to the appropriate runlevel for auto startup? *>>>* > *>>>* >Cheers *>>>* >Jason *>>>* *>>>* On Wed, Feb 1, 2012 at 3:06 PM, Uri Shacked <ushacked at gmail.com< http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> *>>>* wrote: *>>>* *>>>* >* Hi again.... *>>>* *>* *>>>* *>* i installed kamailio 3.2.2 today. *>>>* *>* i used the chkconfig tool to get kamailio run at boot. *>>>* *>* i edited both /etc/init.d/kamailio and /etc/default/kamailio files *>>>* to run *>>>* *>* with 2048Mg of memory. *>>>* *>* now, when i run /etc/init.d/kamailio start, it starts with the right *>>>* *>* amount of memory. *>>>* *>* when i rum "kamailio" only, it starts with 32Mb.... *>>>* *>* *>>>* *>* i use centos 6.2 64bit. *>>>* *>* any ideas? *>>>* *>* *>>>* *>* *>>>* *>* *>>>* *>* *>>>* *>* *>>>* *>* On Tue, Oct 25, 2011 at 1:18 PM, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> *>>>* wrote: *>>>* *>* *>>>* *>>* Hi. *>>>* *>>* I already checked it all. The kamailio start and stop command did *>>>* not *>>>* *>>* work. The init script i used was for debian. Now i use another *>>>* script, *>>>* *>>* for centos. It works. I add the kamailio proccess with chkconfig *>>>* tool. *>>>* *>>* Thanks. *>>>* *>>* *>>>* *>>* On 10/25/11, Sammy Govind <govoiper at gmail.com< http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> *>>>* wrote: *>>>* *>>* > Thanks Alex, *>>>* *>>* > *>>>* *>>* > Hey Uri, can you check if you've /etc/init.d/kamailio file *>>>* exists - open *>>>* *>>* > that file if exists and cross verify few parameters. *>>>* *>>* > Few of these are as follows *>>>* *>>* > *>>>* *>>* > PATH=/sbin:/bin:/usr/sbin:/usr/bin *>>>* *>>* > DAEMON=*/usr/local/sbin/kamailio* *>>>* *>>* > NAME=kamailio *>>>* *>>* > DESC=kamailio *>>>* *>>* > HOMEDIR=/var/run/ *>>>* *>>* > *PIDFILE*=$HOMEDIR/$NAME.pid *>>>* *>>* > DEFAULTS=*/etc/default/kamailio.default* *>>>* *>>* > RUN_KAMAILIO=yes *>>>* *>>* > *>>>* *>>* > Make sure /etc/default/kamailio.default file exits. here are the *>>>* *>>* contents of *>>>* *>>* > my file. *>>>* *>>* > *>>>* *>>* > RUN_KAMAILIO=yes *>>>* *>>* > USER=kamailio *>>>* *>>* > GROUP=kamailio *>>>* *>>* > MEMORY=64 *>>>* *>>* > DUMP_CORE=no *>>>* *>>* > *>>>* *>>* > On Tue, Oct 25, 2011 at 3:38 AM, Klaus Darilion < *>>>* *>>* > klaus.mailinglists at pernau.at< http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> *>>>* wrote: *>>>* *>>* > *>>>* *>>* >> Test the startupscript if they work: *>>>* *>>* >> *>>>* *>>* >> /etc/init.d/kamailio start *>>>* *>>* >> /etc/init.d/kamailio stop *>>>* *>>* >> *>>>* *>>* >> If this works, then make sure to have proper symlinks from the *>>>* *>>* >> corresponding runlevel to the startup script. Do it manually or *>>>* use the *>>>* *>>* >> chkconfig tool. *>>>* *>>* >> *>>>* *>>* >> klaus *>>>* *>>* >> *>>>* *>>* >> *>>>* *>>* >> On 24.10.2011 08:00, Uri Shacked wrote: *>>>* *>>* >> *>>>* *>>* >>> hi again, *>>>* *>>* >>> i found out that on my centos 5.4 the command *>>>* start-stop-daemon that *>>>* *>>* the *>>>* *>>* >>> /etc/init.d/kamailio uses is not recognized. *>>>* *>>* >>> it is kamailio3.2 any ideas? *>>>* *>>* >>> *>>>* *>>* >>> *>>>* *>>* >>> On Mon, Oct 24, 2011 at 3:03 PM, Uri Shacked <ushacked at *>>>* gmail.com< http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> *>>>* *>>* >>> <mailto:ushacked at gmail.com< http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>> *>>>* wrote: *>>>* *>>* >>> *>>>* *>>* >>> Hi, *>>>* *>>* >>> what is the best way to start kamailio at boot? *>>>* *>>* >>> the example in the install is not working for me.... *>>>* *>>* >>> *>>>* *>>* >>> *>>>* * *>>>* *>>* *>>* This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer *>>* *>>* *>* *

2 2

sipp issues with dialog module
by Uri Shacked 08 Feb '12

08 Feb '12

Hi, anyone knows why the dialog module leaves active dialogs even after a bye was recieved when using sipp? once i had issues with the record route and i forced sipp to deal with that, is there something to set on the bye sent by sipp? BR, Uri

2 2

Accessing indexed header PV from app_lua module in Kamailio
by Chris Austin 08 Feb '12

08 Feb '12

Hi all, I have have been attempting to implement Remote-Party-ID support using the app_lua module in Kamailio. Since more than one Remote-Party-ID header may exist, this requires accessing the header pseudo-variable with an index. For example: $(hdr(Remote-Party-ID)[2]) Doing this causes Kamailio to output an error to the log when it fails to parse the PV: 2012-02-02 16:38:58|E| 6(26940) ERROR: app_lua [app_lua_sr.c:892]: invalid pv [$(hdr(remote-party-id)[0])] (-1/26) This was triggered using sr.pv.get("$(hdr(remote-party-id)[0])") from within Lua. I have also tried sr.pv.get("$hdr(remote-party-id)[0]") which also failed: 2012-02-02 18:26:28|E|15(26956) ERROR: app_lua [app_lua_sr.c:892]: invalid pv [$hdr(remote-party-id)[0]] (21/24) Am I doing something wrong? If not, any ideas for a workaround? Thanks for your time.

2 1

Store auth username in registrations
by Spencer Thomason 08 Feb '12

08 Feb '12

Hello, I have clients which register to Kamailio using an auth username. The from username is used for caller id on an INVITE. This causes a problem if the client does not use the auth username in the from for the REGISTER. The caller id number is then saved as the AOR username which doesn't correspond to a valid account. Is there a way I can store the authorized username instead of the From username in the AOR? Thanks, Spencer

2 5

debuger, debug level and log
by Uri Shacked 08 Feb '12

08 Feb '12

hi, i didn't fully understand the differences between the debug level set at the Global Parameters section and the debugger module. i would like to start kamailio with debug level 0, and to be able to raise or lower the level in command line. the log is set to be writen to /var/log/kamailio, and i can trace it using "tail". so, anyideas how to use the debug options with the option to change level at runtime? BR, Uri

2 1

High Availability for ongoing calls
by Mariana Arduini 07 Feb '12

07 Feb '12

Hi guys! I'm facing some problems with dialog information while trying to implement a HA environment. We have 2 clustered SIP servers working as a Active/Standby failover and the use of a Virtual IP address. We want to make sure that if proxy #1 goes down during a call, proxy #2 would be able to handle the sequential requests for that call, since both proxies are using the same database for dialog module information and we have set the db_mode dialog parameter to 1 (REALTIME - any dialog information changes will be reflected into the database immediately). Instead, what happened is proxy #2 was not aware of the dialog and gave us a "500 Server error occurred" message after receiving the BYE. Here are the log errors: /usr/local/sbin/opensips[12617]: New request: BYE /usr/local/sbin/opensips[12617]: The request has TO tag /usr/local/sbin/opensips[12617]: cannot match request to a dialog /usr/local/sbin/opensips[12617]: ERROR:dialog:w_validate_dialog: null dialog /usr/local/sbin/opensips[12617]: in-dialog bogus request /usr/local/sbin/opensips[12617]: validate_dialog return: -4 /usr/local/sbin/opensips[12617]: loose_route() performed, rr headers were decoded /usr/local/sbin/opensips[12617]: ERROR:core:parse_uri: uri too short: <0> (1) /usr/local/sbin/opensips[12617]: ERROR:core:parse_sip_msg_uri: bad uri <0> /usr/local/sbin/opensips[12617]: ERROR:tm:new_t: uri invalid /usr/local/sbin/opensips[12617]: ERROR:tm:t_newtran: new_t failed /usr/local/sbin/opensips[12617]: Could not execute t_relay() Virtual IP address is working, a new call goes through with no problem. Apparently, the dialog module is not saving all dialogs information to the database right away. Does anyone know about any other info we should be storing in the database, using some sort of module parameter or exported function from config file? In other words, what do I need to do in order to send an initial INVITE to one server and, when that server goes down, send the BYE to another server with the same call dialog? Thanks in advance. Mariana.

2 2

FreeBSD comiling problem
by Serhat AKCA 07 Feb '12

07 Feb '12

Hello I want to run Kamailio under FreeBSD. I compiled with the makefiles under freebsd packages. But It gives alotof errors. I want to use 3.2.2 version but I cannot compile. Could you help me To compile in FreeBSD ? Another question, which is the best OS for Kamailio ? In the forums, they are saying FreeBSD is more stable and More performans. Regards Serhat

4 3

Proxy topology question
by Greg Mannie 06 Feb '12

06 Feb '12

Hello After much reading I have come to the realization that after years of using Asterisk I know very little about Sip. I have my Kamailio box up, I have Asterisk 1.8.x running with realtime working. I thought it would be just a case of registering SIP trunks from my provider to the kamailio and registering our internal asterisk servers to the kamailio. Much of what I read talks about using Asterisk as the PSTN interface, but that interface is through a sip trunk purchased from a provider. Won't Kamailio be the PSTN gateway? The idea here is to pool all the sip trunks from the various hosted asterisk solutions (VM running asterisk) and point them all to a proxy to facilitate the aggregation of traffic. I have been reading SIP tutorials and the mailing list archives. If anyone has a sample config and perhaps a little direction it would be highly appreciated. Thank you Greg

2 5

loading carrierroute failes using another DB
by Uri Shacked 06 Feb '12

06 Feb '12

Hi, I changed the location of the DB kamailio works with. i copied the relevalt tables and changed the relevant names in the cfg file. while trying to load kamailio i realized that it misses the "veresion" table. first - can someone explain the version table usage? second, after creating and copying that data from the version table. i tried to load kamailio again and got the follwing: " DEBUG: carrierroute [cr_db.c:345]: loading, cycle 0 0(24689) ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe" any help? thanks, Uri

2 3

"kamctl cr show" when carrierroute table has another name
by Uri Shacked 06 Feb '12

06 Feb '12

Hi, after resolving the field type issue :-), i changed the params in kamctlrc to point to the right DB. now, i notice that when i use the command "kamctl cr show" iget an error that it does not find the table "carrierroute". in the cfg file it is all set and works fine, cr dump is working as expected as well. in the kamctl file i can find the param - CARRIERROUTE_TABLE. but where is it set from? anyone knows where do i make tha change in order for the kamctl to "know" the names of the tables i use? thanks, Uri

1 0

Kamailio 3.1.5 core dump from the snmpstats module
by Klaus Feichtinger 06 Feb '12

06 Feb '12

Hello list, I`ve found a negative phenomenon during startup of kamailio 3.1.x (orgininally found in version 3.1.5) that results in a core dump, triggered by the snmpstats module. During startup of the SIP proxy server machine the linux service "heartbeat" is responsible for starting required ressources like mysql, kamailio and snmpd. These services are monitored by the tool "monit". Monit is supervising the state of the named services and if kamailio is not up and running, monit is trying to restart the service. During startup the mysql database takes some time until it is ready to use and therefore kamailio can`t connect to the database. Monit is detecting this and restarting kamailio. This is done so often until kamailio is up and running. Finally all services are up and running and the machine is working fine. However, I do not like the core dump that was created during this "restart phase". As soon as the module "snmpstats" is not used, no core dump is created any more. Information about the core dump looks as follows: (1) output of kamailio log files: <prefix> kamailio: INFO: <core> [tcp_main.c:4726]: init_tcp: using epoll_lt as the io watch method (auto detected) <prefix> /usr/sbin/kamailio[3027]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) <prefix> /usr/sbin/kamailio[3027]: ERROR: <core> [db.c:289]: could not add connection to the pool <prefix> /usr/sbin/kamailio[3027]: ERROR: lcr [lcr_mod.c:359]: unable to connect to the database <prefix> /usr/sbin/kamailio[3027]: ERROR: lcr [lcr_mod.c:571]: unable to open database connection <prefix> /usr/sbin/kamailio[3027]: ERROR: <core> [sr_module.c:875]: init_mod(): Error while initializing module lcr (/usr/lib/kamailio/modules/lcr.so) <prefix> /usr/sbin/kamailio[3027]: INFO: snmpstats [snmpstats.c:387]: The SNMPStats module got the kill signal <prefix> kamailio: ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe <prefix> kamailio: INFO: <core> [tcp_main.c:4726]: init_tcp: using epoll_lt as the io watch method (auto detected) <prefix> kamailio: WARNING: <core> [daemonize.c:352]: pid file contains old pid, replacing pid <prefix> /usr/sbin/kamailio[3322]: INFO: usrloc [hslot.c:53]: locks array size 512 <prefix> /usr/sbin/kamailio[3322]: INFO: auth [auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled) support disabled <prefix> /usr/sbin/kamailio[3322]: INFO: pua [pua.c:360]: the query returned no result <prefix> /usr/sbin/kamailio[3322]: INFO: pike [ip_tree.c:88]: probing 256 set size <prefix> /usr/sbin/kamailio[3322]: INFO: <core> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 111616 <prefix> /usr/sbin/kamailio[3322]: INFO: <core> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142 <prefix> /usr/sbin/kamailio[3329]: INFO: mi_datagram [mi_datagram.c:326]: a new child 0/3329 (2) output of GDB / core file: Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 1024 -u kamailio -g kama'. Program terminated with signal 11, Segmentation fault. [New process 3027] #0 0xb775539c in freeInterprocessBuffer () at interprocess_buffer.c:354 354 interprocess_buffer.c: No such file or directory. in interprocess_buffer.c (gdb) I know that it is not a fine method in killing kamailio during initialisation. But I have not found a better way yet. The other modules like LCR have no problem with the kill signal during initialisation. My main question is now: how could these core dumps being avoided? Did anybody have the same experience as me? Thanks in advance, Klaus

2 2

Kamailio, Freeradius and CDRTool
by Reda Aouad 05 Feb '12

05 Feb '12

Hi, I am trying to get Kamailio to work with CDRTool. In the step of installing freeradius and adding kamailio's dictionary dictionary.kamailio to freeradius, I get the following error when starting freeradius -X, which fails to start: Error: Errors reading dictionary: dict_init: No ATTRIBUTE "Sip-Method" defined for VALUE "Other" Replacing Kamailio's dictionary with OpenSips' one, freeradius starts correctly. I assume then that there's something missing in Kamailio's dictionary. I am using Kamailio v3.2.0 and it's dictionary file. Any help ? Thanks :) RA

1 1

3.2.2
by dotnetdub 03 Feb '12

03 Feb '12

Hi All, I've just installed 3.2.2. I'm using a config that was working fine with 3.1 and all seems ok. I get one error that I don't get on 3.1 at startup. kamailio: ERROR: <core> [ppcfg.c:79]: ### added subst expression: /DB/mysql This is repeated for all SUBST statements in the config. I get these errors also: [1293000.484026] INFO: task kamailio:19912 blocked for more than 120 seconds. [1293000.484081] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [1293000.484169] kamailio D c4608e58 0 19912 19899 0x00000000 [1293000.484173] f6b5d980 00000082 40827cc6 c4608e58 000006cf c1441e20 c1441e20 000000fd [1293000.484177] f6b5db3c c4608e20 00000000 4085cc46 00049753 ffe3a684 ffffffff f7077bac [1293000.484180] f6b5d9ac f6b5db3c f6b5d980 c102b2b7 00000001 c4608e20 00000004 00000005 [1293000.484184] Call Trace: [1293000.484191] [<c102b2b7>] ? check_preempt_wakeup+0x18c/0x202 [1293000.484195] [<c127f55f>] ? __mutex_lock_common+0xe8/0x13b [1293000.484198] [<c127f5c1>] ? __mutex_lock_slowpath+0xf/0x11 [1293000.484200] [<c127f652>] ? mutex_lock+0x17/0x24 [1293000.484202] [<c127f652>] ? mutex_lock+0x17/0x24 [1293000.484205] [<c123f68b>] ? unix_dgram_recvmsg+0x49/0x1e4 [1293000.484207] [<c10268bd>] ? __wake_up_common+0x34/0x59 [1293000.484210] [<c102a97c>] ? __wake_up_sync_key+0x33/0x49 [1293000.484213] [<c11dd3b0>] ? __sock_recvmsg+0x50/0x58 [1293000.484215] [<c11dd8e7>] ? sock_recvmsg+0x9d/0xb6 [1293000.484218] [<c104a652>] ? autoremove_wake_function+0x0/0x2d [1293000.484221] [<c1092950>] ? __alloc_pages_nodemask+0xf3/0x4d9 [1293000.484224] [<c108e16b>] ? find_lock_page+0x1b/0x36 [1293000.484227] [<c1025769>] ? kmap_atomic_prot+0xd7/0xfc [1293000.484230] [<c113cd18>] ? cpumask_any_but+0x20/0x2b [1293000.484232] [<c11de830>] ? sys_recvfrom+0xc0/0x120 [1293000.484235] [<c1143b63>] ? copy_from_user+0x27/0x10e [1293000.484237] [<c1025769>] ? kmap_atomic_prot+0xd7/0xfc [1293000.484240] [<c10a2deb>] ? handle_mm_fault+0x982/0xa22 [1293000.484243] [<c104a652>] ? autoremove_wake_function+0x0/0x2d [1293000.484246] [<c10baa1d>] ? fsnotify_access+0x5a/0x61 [1293000.484248] [<c11decee>] ? sys_socketcall+0x123/0x1aa [1293000.484251] [<c100813b>] ? sysenter_do_call+0x12/0x28 19912 is MI datagram. I have been getting these errors since 1.x and what I normally do is echo 0 > /proc/sys/kernel/hung_task_timeout_secs to get rid of it. It must be a module that I am loading. Is this possible to debug? I can again get rid of the messages but would love to know whats causing it. BR Shena

1 1

question about NAT being behind of a Load Balancer
by Albert Petit 03 Feb '12

03 Feb '12

Hi , Sorry for previous email. Hit send too soon . Good morning, On my system i've a structure similar to: SIP Clients SIP Platform [-------------------] [---------------------------------] S1 @IP2 S2 @IP3 SIP UACs SIP LB ... [NATted and @IP1 non Nattted] SN @IPN SIP LB is based on Kamailio dispatcher and is responsible of distributing the load between different Servers SIP UAS. S1, S2,... SN Servers have a Kamailio Server too which will forward received balanced traffic to appropiate service (Presence , registrar,...) Considering SIP LB and all SIP UAS have Public IP, how NAT from the UACs could be handled on this scenario? Initially I thought NAT should be handled on SIP LB however I would like the most simple possible load balancer with no need to read user locations . Responsible of location users should be S1,S2,..:SN Then, would it be possible to handle NAT of the SIP UACs *from the SIP UAS *with NAT Helper module? What i'm afraid is: If all traffic to S1,S2,..SN comes from SIP LB then nat_helper could think all clients are NATTed because top most via will be SIP LB. What I need, I think, is that NAT Helper decides if there is NAT looking the Via Header added originally by UAC , and not the top most via (which would be SIP LB). Is that possible? If possible how should i configure Nat helper for that? BR Albert

2 3

Re: [SR-Users] sr-users Digest, Vol 81, Issue 7
by Yufei Tao 03 Feb '12

03 Feb '12

Hi I think I got the tcp_crlf_ping and the tcp_keepalive parameters mixed up. So for tcp_crlf_ping, Kamailio will only respond. While for tcp_keepalive parameters, if enabled on both server and client, both will be sending the keepalives, and the receiver will respond. In order to use tcp_crlf_ping, client's app level rather than TCP level would have to send the crlfcrlf (pings) and process the crlf (pongs) in order to detect if Kamailio is up. Is this right? Yufei On 03/02/12 11:00, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. TCP keepalive on server and client (Yufei Tao) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 3 Feb 2012 10:33:11 +0000 > From: Yufei Tao <yufei.tao(a)redembedded.com> > Subject: [SR-Users] TCP keepalive on server and client > To: <sr-users(a)lists.sip-router.org> > Message-ID: <4F2BB7E7.2090601(a)redembedded.com> > Content-Type: text/plain; charset="ISO-8859-1" > > Hi > > I want to use TCP keepalive to detect dead connections between Kamailio > and my SIP clients. I've got some questions about this: > > 1. If I enable it on Kamailio by setting for example: > > tcp_crlf_ping = yes > tcp_keepalive = yes > tcp_keepidle = 60 > tcp_keepintvl = 10 > tcp_keepcnt = 3 > > >From what I read by gooling, that means Kamailio only responds to the > crlfcrlf sent by the client, but can't be configured to send the > crlfcrlf. Is this right? > > 2. On my client, if I use setsockopt to set the corresponding values: > > SO_KEEPALIVE > TCP_KEEPCNT > TCP_KEEPIDLE > TCP_KEEPINTVL > > would it make the client send crlfcrlf? > > The reason I'm confused is, with these settings, if Kamailio can only > respond to crlfcrlf, but not send, then would the client only respond > too, as in 1. and 2., they seem to configure the same set of parameters? > Please help me clarify this. 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 > > > > ------------------------------ > > _______________________________________________ > sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > End of sr-users Digest, Vol 81, Issue 7 > *************************************** -- 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 1

(no subject)
by Albert Petit 03 Feb '12

03 Feb '12

Hi, SIP UAS1 SIP LB <--------------------->

2 1

Kamailio ims / rtpproxy
by trojak＠st.fri.uniza.sk 02 Feb '12

02 Feb '12

Hi, I am trying to make kamailio PCSCF working with rtpproxy (for NAT traversal). I would like to ask, if there is any tutorial for this, or if you can give me any advice. I also have installed rtpproxy package; #!define WITH_NAT, and defined rtpproxy port in config file... But this is not working. I noticed, that in default config file of pcscf is nothing in routing logic with rtpproxy (or it is unnecessary ?), so I try to copy route[RELAY] route[NAT] route[RTPPROXY] modules from kamailio (no pcscf) config file... but also with no success. Can you help me please ? What I am doing wrong ? Do you have any configuration working with rtpproxy ? Best wishes Lubo

2 3

RPMs for v3.2.2
by Daniel-Constantin Mierla 02 Feb '12

02 Feb '12

Hello, for the records, the RPMs for v3.2.2 are available (32b and 64b: CentOS 5 and 6, RedHat 5.0 and 6.0, OpenSUSE 11.4 and 12.1, Fedora 15 and 16): * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] best way to start kamailio at boot
by Jason Penton 01 Feb '12

01 Feb '12

I am not familiar with Centos but I suspect there is another script being run from somewhere. What I would do is move the /etc/init.d/kamailio script somewhere safe (your home directory for eg). Then reboot and see if kamailio still starts - if it does you need to find out whihc script is being called and where it is. If you are not running X by default you are prob in runlevel 3. If you are running X, then it will be 5. check where you runlevel scripts are pointing. e.g. ls -al /etc/rc3.d/SXXkamailio (X = [0-9]) Cheers Jason On Wed, Feb 1, 2012 at 4:26 PM, Uri Shacked <uri.shacked(a)gmail.com> wrote: > Hi, > > when i reboot the machine, it starts with 32Mb and i do not understand > why... > so, does it use a startup file that i am not familiar with? > what is the appropriate runlevel? how do i set it and where? > > thanks, > Uri > > > >Hi Sammy, > > > >When you use the init script, it is starting Kamailio with the correct > >arguments to get the environment you wish (-m 2048 probably). If however, > >you just call kamailio these args are not there. Im not sure why you are > >running with kamailio command directly. Why don't you add your init script > >to the appropriate runlevel for auto startup? > > > >Cheers > >Jason > > On Wed, Feb 1, 2012 at 3:06 PM, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > > >* Hi again.... > *>* > *>* i installed kamailio 3.2.2 today. > *>* i used the chkconfig tool to get kamailio run at boot. > *>* i edited both /etc/init.d/kamailio and /etc/default/kamailio files to > run > *>* with 2048Mg of memory. > *>* now, when i run /etc/init.d/kamailio start, it starts with the right > *>* amount of memory. > *>* when i rum "kamailio" only, it starts with 32Mb.... > *>* > *>* i use centos 6.2 64bit. > *>* any ideas? > *>* > *>* > *>* > *>* > *>* > *>* On Tue, Oct 25, 2011 at 1:18 PM, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > *>* > *>>* Hi. > *>>* I already checked it all. The kamailio start and stop command did not > *>>* work. The init script i used was for debian. Now i use another > script, > *>>* for centos. It works. I add the kamailio proccess with chkconfig > tool. > *>>* Thanks. > *>>* > *>>* On 10/25/11, Sammy Govind <govoiper at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > *>>* > Thanks Alex, > *>>* > > *>>* > Hey Uri, can you check if you've /etc/init.d/kamailio file exists > - open > *>>* > that file if exists and cross verify few parameters. > *>>* > Few of these are as follows > *>>* > > *>>* > PATH=/sbin:/bin:/usr/sbin:/usr/bin > *>>* > DAEMON=*/usr/local/sbin/kamailio* > *>>* > NAME=kamailio > *>>* > DESC=kamailio > *>>* > HOMEDIR=/var/run/ > *>>* > *PIDFILE*=$HOMEDIR/$NAME.pid > *>>* > DEFAULTS=*/etc/default/kamailio.default* > *>>* > RUN_KAMAILIO=yes > *>>* > > *>>* > Make sure /etc/default/kamailio.default file exits. here are the > *>>* contents of > *>>* > my file. > *>>* > > *>>* > RUN_KAMAILIO=yes > *>>* > USER=kamailio > *>>* > GROUP=kamailio > *>>* > MEMORY=64 > *>>* > DUMP_CORE=no > *>>* > > *>>* > On Tue, Oct 25, 2011 at 3:38 AM, Klaus Darilion < > *>>* > klaus.mailinglists at pernau.at<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > *>>* > > *>>* >> Test the startupscript if they work: > *>>* >> > *>>* >> /etc/init.d/kamailio start > *>>* >> /etc/init.d/kamailio stop > *>>* >> > *>>* >> If this works, then make sure to have proper symlinks from the > *>>* >> corresponding runlevel to the startup script. Do it manually or > use the > *>>* >> chkconfig tool. > *>>* >> > *>>* >> klaus > *>>* >> > *>>* >> > *>>* >> On 24.10.2011 08:00, Uri Shacked wrote: > *>>* >> > *>>* >>> hi again, > *>>* >>> i found out that on my centos 5.4 the command start-stop-daemon > that > *>>* the > *>>* >>> /etc/init.d/kamailio uses is not recognized. > *>>* >>> it is kamailio3.2 any ideas? > *>>* >>> > *>>* >>> > *>>* >>> On Mon, Oct 24, 2011 at 3:03 PM, Uri Shacked <ushacked at > gmail.com <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > *>>* >>> <mailto:ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>> > wrote: > *>>* >>> > *>>* >>> Hi, > *>>* >>> what is the best way to start kamailio at boot? > *>>* >>> the example in the install is not working for me.... > *>>* >>> > *>>* >>> > * > This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer

2 2

Re: [SR-Users] best way to start kamailio at boot
by Uri Shacked 01 Feb '12

01 Feb '12

On Wed, Feb 1, 2012 at 4:26 PM, Uri Shacked <uri.shacked(a)gmail.com> wrote: > Hi, > > when i reboot the machine, it starts with 32Mb and i do not understand > why... > so, does it use a startup file that i am not familiar with? > what is the appropriate runlevel? how do i set it and where? > > thanks, > Uri > > > >Hi Sammy, > > > >When you use the init script, it is starting Kamailio with the correct > >arguments to get the environment you wish (-m 2048 probably). If however, > >you just call kamailio these args are not there. Im not sure why you are > >running with kamailio command directly. Why don't you add your init script > >to the appropriate runlevel for auto startup? > > > >Cheers > >Jason > > On Wed, Feb 1, 2012 at 3:06 PM, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > > >* Hi again.... > *>* > *>* i installed kamailio 3.2.2 today. > *>* i used the chkconfig tool to get kamailio run at boot. > *>* i edited both /etc/init.d/kamailio and /etc/default/kamailio files to > run > *>* with 2048Mg of memory. > *>* now, when i run /etc/init.d/kamailio start, it starts with the right > *>* amount of memory. > *>* when i rum "kamailio" only, it starts with 32Mb.... > *>* > *>* i use centos 6.2 64bit. > *>* any ideas? > *>* > *>* > *>* > *>* > *>* > *>* On Tue, Oct 25, 2011 at 1:18 PM, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > *>* > *>>* Hi. > *>>* I already checked it all. The kamailio start and stop command did not > *>>* work. The init script i used was for debian. Now i use another > script, > *>>* for centos. It works. I add the kamailio proccess with chkconfig > tool. > *>>* Thanks. > *>>* > *>>* On 10/25/11, Sammy Govind <govoiper at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > *>>* > Thanks Alex, > *>>* > > *>>* > Hey Uri, can you check if you've /etc/init.d/kamailio file exists > - open > *>>* > that file if exists and cross verify few parameters. > *>>* > Few of these are as follows > *>>* > > *>>* > PATH=/sbin:/bin:/usr/sbin:/usr/bin > *>>* > DAEMON=*/usr/local/sbin/kamailio* > *>>* > NAME=kamailio > *>>* > DESC=kamailio > *>>* > HOMEDIR=/var/run/ > *>>* > *PIDFILE*=$HOMEDIR/$NAME.pid > *>>* > DEFAULTS=*/etc/default/kamailio.default* > *>>* > RUN_KAMAILIO=yes > *>>* > > *>>* > Make sure /etc/default/kamailio.default file exits. here are the > *>>* contents of > *>>* > my file. > *>>* > > *>>* > RUN_KAMAILIO=yes > *>>* > USER=kamailio > *>>* > GROUP=kamailio > *>>* > MEMORY=64 > *>>* > DUMP_CORE=no > *>>* > > *>>* > On Tue, Oct 25, 2011 at 3:38 AM, Klaus Darilion < > *>>* > klaus.mailinglists at pernau.at<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > *>>* > > *>>* >> Test the startupscript if they work: > *>>* >> > *>>* >> /etc/init.d/kamailio start > *>>* >> /etc/init.d/kamailio stop > *>>* >> > *>>* >> If this works, then make sure to have proper symlinks from the > *>>* >> corresponding runlevel to the startup script. Do it manually or > use the > *>>* >> chkconfig tool. > *>>* >> > *>>* >> klaus > *>>* >> > *>>* >> > *>>* >> On 24.10.2011 08:00, Uri Shacked wrote: > *>>* >> > *>>* >>> hi again, > *>>* >>> i found out that on my centos 5.4 the command start-stop-daemon > that > *>>* the > *>>* >>> /etc/init.d/kamailio uses is not recognized. > *>>* >>> it is kamailio3.2 any ideas? > *>>* >>> > *>>* >>> > *>>* >>> On Mon, Oct 24, 2011 at 3:03 PM, Uri Shacked <ushacked at > gmail.com <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > *>>* >>> <mailto:ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>> > wrote: > *>>* >>> > *>>* >>> Hi, > *>>* >>> what is the best way to start kamailio at boot? > *>>* >>> the example in the install is not working for me.... > *>>* >>> > *>>* >>> > * >

1 0

best way to start kamailio at boot
by Uri Shacked 01 Feb '12

01 Feb '12

Hi, what is the best way to start kamailio at boot? the example in the install is not working for me....

5 9

subscriber specific dialplan
by Bayram Karagoz 01 Feb '12

01 Feb '12

Hi, I'm trying to make when each subscriber dials 7 digits add subscriber based specific prefix to dialed number defined in subscriber or dialplan table. I know that it's possible with avp but I'm not professional about avp. could you send me a successful scenario and config example please?

1 1

redirect client to keep original ruri
by Andrew Pogrebennyk 01 Feb '12

01 Feb '12

Hello, assuming that I want to use contact from 302 response as outbound proxy but keep the original Request-URI, what should I do? Calling the revert_uri() after get_redirects() in failure_route doesn't do the trick.

3 7

save a Q850 cause reason from a reply
by Uri Shacked 01 Feb '12

01 Feb '12

Hi guys..... the header searching and saving the Q850 code didn't work. here is what i wrote in the cfg, in the failure route: *# manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_CRIT", "failure \n"); if($avp(CAR)=="CALLOUT"){ xlog("L_CRIT", "failure in callout \n"); ##if(is_present_hf("Reason")) { xlog("L_CRIT", "failure in reason\n"); $avp(Q850)=$(hdr(Reason){param.value,cause}{s.int}); xlog("L_CRIT", "failure in reason $avp(Q850)\n"); ##}* here is the log print...... * 3(3433) CRITICAL: <script>: failure 3(3433) CRITICAL: <script>: failure in callout 3(3433) CRITICAL: <script>: failure in reason 3(3433) ERROR: <core> [lvalue.c:347]: non existing right pvar 3(3433) ERROR: <core> [lvalue.c:407]: assignment failed at pos: (524,15-524,54) 3(3433) CRITICAL: <script>: failure in reason <null>* the command *is_present_hf("Reason")) * is under ## because of tests... when it was in use, the script didn't even went in the "if", the command didn't recognize the header. ** here is the sip trace for the 486, the header is there... * Message Header Call-ID: 6994099017399497785-1328029200-13379 From: <sip:36264529;ss7-cpc=10;cpc=ordinary@172.16.10.3 ;user=phone>;tag=6994099017399497785 To: <sip:39955011@172.16.60.161;user=phone>;tag=9299942026613191738 CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.60.161;received=172.16.60.161;branch=z9hG4bK08d6.c653f346.2,SIP/2.0/UDP 172.16.10.3:5060;branch=z9hG4bK-6110080000017439-ac100a03-1 Supported: timer,100rel Reason: Q.850;cause=17;text="UserBusy" Reason Protocols: Q.850 Cause: 17(0x11)[User busy] Content-Length: 0*

2 1

Re: [SR-Users] Kamailio v3.2.2 Released
by Daniel-Constantin Mierla 01 Feb '12

01 Feb '12

On 2/1/12 11:58 AM, Uri Shacked wrote: > worked OK, thanks. > [...] ok, keep cc-ing the list for topics discussed there, so people reading the archive later know what was the final solution. Cheers, Daniel > > On Wed, Feb 1, 2012 at 11:40 AM, Uri Shacked <ushacked(a)gmail.com > <mailto:ushacked@gmail.com>> wrote: > > i will try again. > i did edit the modules.lst file with all the modules i use.... > will let you know ASAP. > > On Wed, Feb 1, 2012 at 11:33 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > you haven't compiled and installed db_mysql from 3.2.2, so it > is installed the one from 3.2.1. Be sure you include db_mysql > in the list of modules to be compiled and installed, one > option is: > > make FLAVOUR=kamailio include_modules="db_mysql" cfg > make all > make install > > Cheers, > Daniel > > > On 2/1/12 10:30 AM, Uri Shacked wrote: >> >> made install and got the following error: >> "kamailio >> loading modules under >> /usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/ >> 0(21428) ERROR: <core> [sr_module.c:415]: ERROR: module >> version mismatch for >> /usr/local/lib64/kamailio/modules/db_mysql.so; core: >> kamailio 3.2.2 (x86_64/linux); module: kamailio 3.2.1 >> (x86_64/linux)" >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla --http://www.asipto.com <http://www.asipto.com/> > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] Kamailio v3.2.2 Released
by Uri Shacked 01 Feb '12

01 Feb '12

> made install and got the following error: > "kamailio > loading modules under > /usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/ > 0(21428) ERROR: <core> [sr_module.c:415]: ERROR: module version mismatch > for /usr/local/lib64/kamailio/modules/db_mysql.so; core: kamailio 3.2.2 > (x86_64/linux); module: kamailio 3.2.1 (x86_64/linux)" >

3 2

Alternative Debian APT repository
by Daniel-Constantin Mierla 01 Feb '12

01 Feb '12

Hello, besides existing deb.kamailio.org, I added builds for Debian 5.0, 6.0 and Ubuntu 10.04 to the OpenSUSE Build Service (OBS). These repositories are built only for stable releases in 3.2.x series. Main target for them is to be sure the releases compile properly, across main Linux distributions, OBS providing an easy interface to keep updating and building until everything is ok. Anyhow, these alternative repositories may be handy to use for some of you. Details about setting APT sources are available at: http://www.kamailio.org/wiki/packages/debs#using_obs_repository Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] [newbie] questions
by Fred Posner 01 Feb '12

01 Feb '12

On Jan 31, 2012, at 6:32 PM, Me wrote: > Hi, Fred, > > >> For the sake of argument, let's say it was a smart-ass answer-- if your post was to "be pointed in the right direction where I could get (further) help" then that next help should either be: >> >> a) the wiki >> b) a training class >> c) a consultant >> > >> I'd say it again though - this thread is for me to seek answers/advice/help/guidance - if you, the previous poster, or anyone else for that matter is unwilling or unable to provide one, then just move along - there is nothing to see here. Smart-arse comments like "get a consultant" isn't what I am looking for, nor is the reason for starting this thread on this mailing list. >> > Can you read? > >> On forums I notice that those who get a better response usually use their real name instead of "Me" with a private/alias email address. >> > If I tell you that my real name is Josh Oliveira 'Capo' De Souza would that make you feel any better? Would you like to know my ssn as well? How about my date of birth? > > If I tell you where I live would that make you go away and prevent you from posting off-topic drivel on the above thread or ask silly questions you know you won't get answers to? TA! > > Josh Josh, I'm sorry-- my response wasn't insulting whatsoever. I'm not sure what your problem is, but to me it looks like you're looking for a fight. I erred on the side of thinking you were above that-- my mistake. I bid you a good day. --fred

1 0

[newbie] questions
by Me 31 Jan '12

31 Jan '12

Apologies if any of the questions below are a bit dumb - I don't pretend to be an expert in SIP/VOIP - I am just an ordinary user looking for answers. Our current setup involves processing a small number of internal sip accounts (up to 10, no more than that) and one "public" one (with a separate registrar) in the following way: On our server we have three interfaces: eth0, eth1 and tun0. eth0 is our entry point to the public internet, eth1 faces our internal network and tun0 is a private vpn, which connects all our smartphones to the internal network (via Wifi, EDGE/2G/3G etc). This gives us the mobility. Up until now, we have been routing voip calls via a commercial (closed source), very limited, terribly outdated (Pentium code base!) and rather buggy sip proxy. I had to employ a lot of hacks on our server in order to route calls as this proxy can only listen on a single interface. It was also a nightmare to maintain. Unsurprisingly, I decided that enough is enough and I am now determined to replace it. We route calls in the following way: all machines (PCs are all Linux based) & smartphones have their own sip/voip client installed on them (also using bluetooth). Internal calls are routed via the proxy between ourselves either on the internal net (eth1), or between the vpn and eth1 (eth1<->tun0). External calls (going out, i.e. outbound) are routed externally to our registrar, using a single separate voip account, via eth1<->eth0 or tun0<->eth0. As I am now looking to replace our proxy, I looked at Kamailio, but was soon completely overwhelmed by it (no offence intended, it was just too much to take at first). I would appreciate if any of you could give me a hand, or at least point me in the right direction, with the following issues: I presume I could configure Kamailio to listen on more than one interface and act as a proxy. How do I do that, so that it listens on all 3 interfaces and proxies requests in the following way: - calls made to <userX>@ourdomain.net to be routed internally via eth1 (internal net) or tun0 (private vpn); - calls made to anybody else to be routed externally via eth0 (public) using the separate "public" sip account with our external registrar; - calls made to the public sip account (from outside - the "public") need to be routed to a "nominated" internall account (say <user0>@ourdomain.net); - all other (internal) calls need to be routed depending on which interface this account has been registered/logged in - either the internal net (eth1) or the private vpn (tun0 - the smartphones). Obviously, calls need to be received (and routed properly) from all 3 interfaces. Is all of this possible with Kamailio? I want to avoid unnecessary complexities of the setup (as I already mentioned above - I am just a user and by no means an expert in sip/voip) and do not want to deploy something I do not need - I need to keep the memory footprint to a bare minimum, possibly without sacrificing performance. Once this is done, I would then move on to the next phase and use IM & ENUM, but this is once the above works. I looked at other alternatives, but I got very confused there as well - I couldn't figure out what exactly is the difference between, say, OpenSER, Kamailio, OpenSIPS and SIP-Router even? What is the best software to use in order to achieve the above setup? One last thing - I am a developer by trade and I am not afraid of "tweaking" things when needed. I was successful in compiling Kamailio from source (I use Fedora on all our machines) and I was pleased that I could exclude from the RPM .spec file the modules I think I did not need. I also made some modification of my own to make the database modules (mysql, postgresql and unixodbc) configurable in the same way the rest of the modules are. I could submit patches, if needed, so that these are incorporated into future releases - how do I do that? I could not do the same with OpenSIPS, however (which I also tried - out of curiosity!) - everything there seems to be lumped and compiled together regardless of whether it is needed or not. Any help as to helping me with the above issues is greatly appreciated, many thanks in advance for taking the time!

7 13

Kamailio Proxy and Asterisk
by Greg Mannie 31 Jan '12

31 Jan '12

Hello, We have been using Asterisk for sometime and over the last year have started hosting instances for our clients on a vmware platform. These virtual pbx are located on public ip addresses and each customer has their own SIP trunk arrangements with various providers. We have decided to pool our resources and would like to start aggregating traffic. From reading online I have installed Kamailio 3.1.x with Asterisk 1.6.x. On top of this I have installed Siremis 3.2 in hopes of using it as a graphical front end. I am wanting to use Kamailio as a proxy to the SIP providers and allow the Asterisk to register only with Kamailio. Is there a link to some example configs with a tutorial pertaining to this type of deployment? If this is a good solution, it is my intention to recommend a consultant be hired to assist with a production server. I just need to learn allot more, before I can recommend Kamailio for the job. Do you think it's a good fit? Thank you for your kind responses, Greg

3 2

Enable application/pidf+xml
by Vogt, Jonathan 31 Jan '12

31 Jan '12

Hello all, I'm trying to set up kamailio as a sip server with presence support for Bravis. So far I got the call processing running on either openser-1.3.2-3build1 or 3.2.0+lucid1 running. Problem is the notification is not using Content-Type: application/pidf+xml My openser config is at http://pastebin.ca/2106401 My kamailio config is pretty much a stock configuration. It is at http://pastebin.ca/2106404 I enabled mysql and presence. I also tried using x-lite but there is no presence information either. modparam("presence_xml", "force_active", 1) did not seem to make a difference. (I cleaned my watchers table) I can see the database has the information but the notify packets are just like this: NOTIFY sip:2001@10.0.76.0:56239 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.23;branch=z9hG4bKcb76.b2dab2d4.0 To: sip:2001@sip.intern.domain;tag=90cd05fe2b0619108127da16a6e721e1 From: sip:2002@sip.intern.domain;tag=10.785.1327490724.1 CSeq: 22 NOTIFY Call-ID: 90cd05fe-2b06-1910-9660-da16a6e721e1 Content-Length: 0 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Max-Forwards: 70 Event: presence Contact: <sip:10.0.10.23:5060> Subscription-State: active;expires=300 Where is my mistake. As I'm new to kamailio please bear with me, if I just didn't find the right switch. Thanks Jonathan Vogt

4 13

IRC devel meeting
by Daniel-Constantin Mierla 31 Jan '12

31 Jan '12

Hello, I would like to schedule an IRC meeting to discuss about the development of the project in 2012. The topics coming in my mind at this moment: - roadmap to next major release v3.3.0 - outstanding issues at this time, if any - packaging - ways to automate the process better - documentation - suggestions to improve it and attract more contributions in this field - ideas for new features - administration - web site, mailing lists, etc ... do we miss some good resource out there that should be added in the eco-system? Please add your suggestions for agenda to the wiki page: * http://www.kamailio.org/wiki/devel/irc-meetings/2012a I proposed two dates (Jan 26 and Jan 31, 15:00GMT) and made a Doodle poll for it: * http://www.doodle.com/qzteweb62sqax5r6 If you wish other options for dates, let me know. Based on results, we will announce the final date. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 3

Upgrade Kamailio from V3.1.5 to V3.2
by Phillman25 Kyriacou 31 Jan '12

31 Jan '12

Dear List I have Kamailio v3.1.5 currently installed and working, as this is currently in production, could someone please tell me what would be the safest way to upgrade to v3.2 without losing any data? Thanking you in advance for your support Phillip

1 0

new git commiter: Claudio Furrer
by Henning Westerholt 31 Jan '12

31 Jan '12

Hello, Claudio Furrer <elcaio(a)gmail.com> has now git commit access for his work on gentoo packaging. He create several patches and extensions for our gentoo package files in the last months and also maintain an own unofficial portage overlay tree at http://gpo.zugaina.org/net-misc/sip-router since some time. He just updated the gentoo packages in git master today, so if you're an gentoo user why not just give it a try - he is looking forward to your feedback. Welcome Claudio! Best regards, Henning

1 0

panning for v3.2.2
by Daniel-Constantin Mierla 31 Jan '12

31 Jan '12

Hello, I would like to package 3.2.2 on the same day with IRC devel meeting announced in the previous email (if is is happening this month) -- it is getting close to 2 months since v3.2.1. If you are aware of issues, report them on bug tracker to see what can be done before 3.2.1. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 2

xmlops/xpath access of multiple nodes
by Alex Balashov 30 Jan '12

30 Jan '12

Hi, I guess this is more of an xpath question than a Kamailio one, so forgive me. All the same, how does one deal with a document like this using $xml(...): <?xml version="1.0" encoding="utf-8" ?> <sheep> <lamb> ... </lamb> <lamb> ... </lamb> </sheep> Specifically, how do I access the second, third, Nth element with the same tag, but without knowledge of any unique identifying attributes? Is there a useful way to iterate through them in Kamailio route script? I would have thought something like this could work: $var(i) = 0; while(defined $xml(x=>xpath:/sheep/lamb[$var(i)])) { ... $var(i) = $var(i) + 1; } However, obviously, that's not it. So what's the right way to do it? My desire is to buffer data from these nodes into an AVP array. As you might guess, it's a route set over which I need to serially fork. Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 1

Re: [SR-Users] save a Q850 cause reason from a reply
by Uri Shacked 30 Jan '12

30 Jan '12

On Mon, Jan 30, 2012 at 7:39 PM, Uri Shacked <uri.shacked(a)gmail.com> wrote: > Hi, > thanks for the options. i will test it soon. > looks like a perfect solution. > > BR, > Uri >

1 0

Kamailio Crashing
by Ali Jawad 30 Jan '12

30 Jan '12

Hi I am running Kamailio and at times it just crashes, sometimes it happens once a day sometimes a few times a day, in debug mode it shows the following when it crashes http://pastebin.com/ZcQRnXmM GDB shows http://pastebin.com/02wVK0eZ GDB BT Shows http://pastebin.com/3TBAuQAp Please advice -- *Ali Jawad * *Information Systems Manager* *Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +9611373725/ext 116 FAX: +9611375554*

2 4

alias problem
by Stoyan Mihaylov 30 Jan '12

30 Jan '12

My whole configuration is: [Sip clients] < = > Kamailio 3.2 <=> Asterisk servers (behind Kamailio) Asterisk servers have only local IP addresses, and I use t_relay instead of forward. Kamailio runs on same server as rtpproxy. Everything is fine if clients connect to Kamailio with its IP address - global, or if they are behind Kamailio with local address. When clients connect to Kamailio using sip.ourcompany.com, then call (video also) is OK, but ACK and BYE do not work. BYE receives not here (404), and ACK die somewhere. I forward BYE and ACK in case when src_ip==$td to Asterisk server. If one of clients use IP - then calls initiated from it are OK (BYE/ACK - are going correctly - to Asterisk and to other client also). But calls from other client have problems with BYE and ACK. To use sip.ourcompany.com - I put: alias=sip.ourcompany.com route[ACKBYE] { #!ifdef WITH_PSTN if (is_method("BYE|ACK")) { xlog("L_ALERT","AB $rm $sht(forw=>$ft) $td"); if(src_ip==$td){ #I have to rewrite du - messages loop in Kamailio, I store in $sht(forw=>$ft) $du which I use during INVITE. $du=$sht(forw=>$ft); route(RELAY); exit; } xlog("L_ALERT","ACK,Bye Not me"); } #!endif return; }

2 2

Open Source billing
by Abid Naseem 30 Jan '12

30 Jan '12

Looking for open source billing solution for Kamailio: 1) rating 2) client interface for cdr 3) admin interface Help needed. Thanks and keep inspiring! -abid Sent from my iPhone

3 4

exec_avp size limitation
by Gustavo Garcia Bernardo 30 Jan '12

30 Jan '12

Hi folks, I'm having problems with exec_avp because is truncating the output of the script executed. Looking at the code I see exec_avp is imposing a limit of MAX_URI_SIZE (1024) but I don't understand why. The limit should be the maximum size for an avp variable in my opinion. Regards, G. Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at http://www.tid.es/ES/PAGINAS/disclaimer.aspx

2 1

memory leak fix on xcap_ auth.c
by laura testi 30 Jan '12

30 Jan '12

Hi Daniel, I'm sending you some memory leak fix we have found and fixed on modules/utils/xcap_ auth.c. The fixed memory leak is in pres_watcher_allowed (), and it is about the "xmlFreeDoc" that it's never called for the "xcap_tree" variable. We downloaded the source file from master branch and then we fixed it. You can find here the diff from original and new code and in attach both the original and the new files. diff -u a/modules/utils/xcap_auth.c b/modules/utils/xcap_auth.c --- a/modules/utils/xcap_auth.c +++ b/modules/utils/xcap_auth.c @@ -280,13 +280,15 @@ } node= get_rule_node(subs, xcap_tree); - if (node== NULL) + if (node== NULL){ + xmlFreeDoc(xcap_tree); return 0; - + } /* process actions */ actions_node = xmlNodeGetChildByName(node, "actions"); if (actions_node == NULL) { LM_DBG("actions_node NULL\n"); + xmlFreeDoc(xcap_tree); return 0; } LM_DBG("actions_node->name= %s\n", actions_node->name); @@ -294,6 +296,7 @@ sub_handling_node = xmlNodeGetChildByName(actions_node, "sub-handling"); if (sub_handling_node== NULL) { LM_DBG("sub_handling_node NULL\n"); + xmlFreeDoc(xcap_tree); return 0; } sub_handling = (char*)xmlNodeGetContent(sub_handling_node); @@ -302,6 +305,7 @@ if (sub_handling == NULL) { LM_ERR("Couldn't get sub-handling content\n"); + xmlFreeDoc(xcap_tree); return -1; } if (strncmp((char*)sub_handling, "block", 5) == 0) { @@ -325,10 +329,12 @@ else { LM_ERR("unknown subscription handling action\n"); xmlFree(sub_handling); + xmlFreeDoc(xcap_tree); return -1; } xmlFree(sub_handling); + xmlFreeDoc(xcap_tree); return 0; Kind Regards, laura

2 1

Kamailio TLS with intermediate CA certificates
by Daniel Pocock 30 Jan '12

30 Jan '12

I found that my TLS client was not happy because my server cert is signed by an intermediate root. A quick search in Google found other people mentioning the same problem, but no solution or documentation. I've had a quick look in the Kamailio source and I notice it is using the call: SSL_CTX_use_certificate_chain_file to load the certificate specified in tls.cfg with certificate=myserver.pem In practice, this means the intermediate certificates can be appended to myserver.pem and Kamailio will present them to the TLS client: Example: Trust heirarchy: trusted root - inter 1 - inter 2 - server.example.com.pem Construct the PEM file in this exact order: cat server.example.com.pem > chain-server.example.com.pem cat inter2.pem >> chain-server.example.com.pem cat inter1.pem >> chain-server.example.com.pem and then, in tls.cfg: certificate=chain-server.example.com.pem

4 7

Kamailio for NATted users with ICE and TURN
by Daniel Pocock 30 Jan '12

30 Jan '12

There are now a number of TURN implementations available: http://www.resiprocate.org/ReTurn_Overview http://turnserver.sourceforge.net/ and for many purposes TURN relays and ICE are much better than relying on NAT helper and rtpproxy When clients use ICE: - the client gets upset if Kamailio mangles the SDP to use rtpproxy - it is actually not necessary for the SDP to have addresses tweaked - because ICE deals with it anyway In practice, it still appears necessary for the registration server to deal with NATted contact addresses though Can anyone clarify how Kamailio should be configured for ICE clients? In other words, is it possible to have Kamailio dealing with NATted contact addresses, but not touching the SDP? To limit the scope of the question: - assume non-ICE clients do not need to be supported at all (just ignore them, although a nice cfg would scan all SDP and send a helpful error back if it sees a 192.168 address and no a=ice* attributes) - assume clients are connecting with TLS over TCP, so there is no risk that some intermediate device like a router is mangling the SIP headers or SDP

2 2

cr_dump_routes buffer too small
by Mino Haluz 30 Jan '12

30 Jan '12

Is there any way how to print it this way ? sercmd -s unixs:/tmp/kamailio_ctl mi_fifo cr_dump_routes error: 500 - Internal server error processing 's': buffer too small (overflow) (-2)

1 0

RPM packages - repositories update
by Daniel-Constantin Mierla 30 Jan '12

30 Jan '12

Hello, the repositories with build RPMs done via OpenSUSE Build Services (OBS) were updated. Fedora 14 is no longer offered by OBS, instead I added support for Fedora 15 and 16. Also, there are now dedicated builds for RedHat 6 and OpenSUSE 12.1. Links to repos are provided at: * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

save a Q850 cause reason from a reply
by Uri Shacked 30 Jan '12

30 Jan '12

Hi, the gateway i use for sending call to pstn has an option of sending a header with Q850 reason. it looks like this: " Reason: Q.850;cause=17;text="UserBusy" Reason Protocols: Q.850 Cause: 17(0x11)[User busy] " now, i need to save the reason for reports. the reason changes and i am trying to find a way to parse it and save the code only. usualy i use the "search" function to get text from headers that are the gateway's inventions :-) here i can find the text "Q850....." but how do i get the cause code? i guess i cab write something long with textops and parsing. but, is there a better and nicer way to perform it? thanks, Uri

3 2

Registrar ignoring socket
by Spencer Thomason 30 Jan '12

30 Jan '12

Hello, I'm working on building a HA setup with separate registrar and proxy servers. The registrar server writes to a master database that is replicated to a local database on each of the proxy servers. A load balancer then sits in front of the proxys. When routing to a registered user on the proxys I receive an error message that the socket is unknown and ignored (obviously because its on another machine). Is this just a warning message or are there any consequences to doing it this way? Is my topology flawed and would there be a better way? Thanks for your input, Spencer Connected by DROID on Verizon Wireless

3 3

Re: [SR-Users] Kamailio Redundancy examples/pointers required
by Javier Gallart 30 Jan '12

30 Jan '12

Hello the net.ipv4.ip_nonlocal_bind = 1 options works fine. We have a very basic redundancy level, but it fits our current needs; sometimes it's even harder to troubleshoot the redundancy failures than the actual failures. We have 2 identical servers running kamailio and listening to the same ip addresses/ports (you need to enable the sysctl option for that). A heartbeat process controls in what machine the actual ip addresses are active. The DB is in one of the servers but we update the shared memory in both. For us having the DB down for a while is not critical because all the data is in memory. When one servers goes down the other one takes over immediately. It's a very comfortable setup also for doing upgrades in the configuration in a controlled manner. Hope it helps Regards Javi > ------------------------------ > > Message: 6 > Date: Thu, 26 Jan 2012 12:13:18 +0500 > From: Sammy Govind <govoiper(a)gmail.com> > Subject: Re: [SR-Users] Kamailio Redundancy examples/pointers required > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAJUJwtjMGTuNe1Cu5+BDthgj0xrSJrBM03XzcjZhWJ8Ar7=onQ(a)mail.gmail.com > > > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > Thanks for taking out time for this. really appreciate your ideas. Here are > few things I've found useful so far. > > http://kamailio.org/events/2011-fosdem/p_usrloc.pdf > This looks like one thing that one could have to ensure more reliability > but we do need other application level and physical level failover > techniques additionally. > > For using any technique like Linux-HA, Heartbeat, keepalived, UCARP or any > floating/virtual IP techniques I really hope that the sysctl option > mentioned in b/m thread really works. > http://www.mail-archive.com/sr-users@lists.sip-router.org/msg03714.html > http://old.nabble.com/Kamilio-Failover-Options-td21947502.html > DNS SRV is one option. Using another Kamailio on top of two Kamailio is > again a SPOF so I think its of no use at all. > > I hope to get a very reliable and working solution out of these forums. > > Regards, > Sammy > > On Wed, Jan 25, 2012 at 9:49 PM, Andreas Granig <agranig(a)sipwise.com> > wrote: > > > Hi Sammy, > > > > On 01/25/2012 02:08 PM, Sammy Govind wrote: > > > I've been looking for ways to create a redundant Kamailio cluster. I've > > > googled alot but haven't got any concrete or final wording on any one > > > solution that'll just work perfectly. The basic requirement is that in > > > > HA is never really easy :) > > > > > case of Kamailio application failure or in case of physical machine > > > error the whole setup just starts working on secondary server. > > > > > > Please suggest anything whichever you guys are using. Any reference > URLs > > > would be very much appreciated. > > > > For 1+1 setups, you can use a master/master replication (people also use > > drbd to sync the whole table space, not sure how good that works) on db > > level, use write-back mode in kamailio to make sure your registrations > > are written to DB immediately, then use heartbeat to do the IP/process > > fail-overs on network/power failures. On top, you can do manual checks > > to trigger a fail-over in case of high-level failures. I guess this is > > the typical and most straight-forward scenario. Your mileage might vary, > > depending on specific environment variables (e.g. you might want to > > announce your active server via RIP/OSPF using quagga instead of the > > more common gratuitous arp if your servers are geographically split). > > > > For scaling larger, put a kamailio instance acting as load-balancer in > > front of your proxies and scale them by providing some sharding > > information to the lb. > > > > Also the new db_cassandra backend in master branch looks pretty > > promising when it comes to HA/scaling. Maybe someone can provide some > > best practices regarding that as well? > > > > Andreas > > > > > > _______________________________________________ > > 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 > > > > >

2 2

Using forward(); with PVs
by Spencer Thomason 28 Jan '12

28 Jan '12

Hello, I'm experiencing some difficulty using forward with PVs. If the IP and port are specified directly everything works but if I use PVs for the values startup fails with a configuration error. What I'm trying to do is statelessly forward an OPTIONS request to a B2BUA that sits behind Kamailio without hard coding the address into the configuration file. Is there a better way to accomplish this? Thanks, Spencer Connected by DROID on Verizon Wireless

2 2

Selects vs PVs
by Spencer Thomason 28 Jan '12

28 Jan '12

Hello all, I'm currently reworking our routing scripts in preparation for an upgrade to 3.2.2 from 3.1.5. I'm taking this time to perform a general cleanup an optimization. In regard to selects vs PVs, is there any reason to choose one or the other select over a PV when both exist for the needed data? Thanks, Spencer

2 1

AVPs vs Variables
by Spencer Thomason 27 Jan '12

27 Jan '12

Hello all, I'm a little confused as to the difference between avps and vars and when I would use one or the other. I'm currently using sqlops to populate a few avps. Would vars be more suitable? Thanks, Spencer Connected by DROID on Verizon Wireless

3 3

pua_usrloc, PUBLISH not send when contact is deleted or expires
by Giacomo Vacca 27 Jan '12

27 Jan '12

Hi all, I've made a clean installation of kamailio 3.2.0 from 'http://deb.kamailio.org/kamailio32 squeeze main' repository, on a squeeze debian machine (2.6.32-5-686). Installed also kamailio-presence-modules, same version. I'm just using the default kamailio.cfg that comes with the package, with these minimal additions: loadmodule "db_text.so" loadmodule "pua.so" loadmodule "pua_usrloc.so" ... modparam("pua", "db_url", "text:///usr/share/kamailio/dbtext/kamailio") modparam("pua_usrloc", "default_domain", "192.168.142.130") Inside the registration management logic I've added pua_set_publish() right before saving the location: route[REGISTRAR] { if (is_method("REGISTER")) { ... pua_set_publish(); if (!save("location")) sl_reply_error(); exit; } } When I first register (XLite 4.1 for Windows), pua_usrloc correctly gets the callback and is able to generate a PUBLISH request through pua. But when there's an un-registration, or when the contact expires, even if the callbacks are correctly fired, the PUBLISH (which I'd expect with status 'closed') is not generated. >From the logs (debug level 4), an un-registration: 8(7652) DEBUG: usrloc [ul_callback.h:89]: contact=0xb50ae7cc, callback type 4/4, id 3 entered 8(7652) DEBUG: pua_usrloc [ul_publish.c:216]:DELETE type 8(7652) DEBUG: pua_usrloc [ul_publish.c:255]: uri= sip:20701090@192.168.142.130 8(7652) DEBUG: pua_usrloc [ul_publish.c:66]: publ: 8(7652) DEBUG: pua_usrloc [ul_publish.c:67]: uri= sip:20701090@192.168.142.130 8(7652) DEBUG: pua_usrloc [ul_publish.c:68]: id=UL_PUBLISH.NmY5Zjc2YjcwNzEwNTk4YTY1NzA2Y2Y0MWQyMGRhOGY. 8(7652) DEBUG: pua_usrloc [ul_publish.c:69]: expires= 0 8(7652) DEBUG: pua [send_publish.c:403]: pres_uri=sip:20701090@192.168.142.130 8(7652) DEBUG: pua [hash.c:121]: core_hash= 444 8(7652) DEBUG: pua [hash.c:174]: record not found 8(7652) DEBUG: pua [send_publish.c:444]: insert type 8(7652) DEBUG: pua [send_publish.c:448]: UPDATE_TYPE and no record found 8(7652) DEBUG: pua [send_publish.c:454]: request for a publish with expires 0 and no record found 8(7652) DEBUG: sl [sl.c:278]: reply in stateless mode (sl) As an additional information, the pua table in /usr/share/kamailio/dbtext/kamailio/pua has not been populated. The AOR was correctly displayed with 'ul show' while the registration was valid. Any suggestion on what am I be doing wrong? Would it be worth testing the same scenario using mysql as DB? Thanks in advance for your time. Giacomo Truphone Limited is a limited liability company registered in England & Wales, registered office: 5 New Street Square, London EC4A 3TW. Registered No. 04187081. VAT No. GB 851 5278 19. Tru is a brand name of Truphone and is a Truphone Communications Service. Truphone is a trading name for a number of distinct legal entities that operate in combination. www.truphone.com<http://www.truphone.com>. This e-mail, and any attachment(s), may contain information which is confidential and/or privileged, and is intended for the addressee only. If you are not the intended recipient, you may not use, disclose, copy or distribute this information in any manner whatsoever. If you have received this e-mail in error, please contact the sender immediately and delete it.

2 11

rpid to from header.
by David 27 Jan '12

27 Jan '12

One of my upstream gateways does not support rpid, so I want to rewrite the from header from the rpid. is just a matter if setting $fU = $rpid? Thanks.

3 3

dbcassandra - 483
by David 27 Jan '12

27 Jan '12

Anca, for some strange reason I am getting "SIP/2.0 483 Too Many Hops" back when I try to register. I am running kamaikio -EEEEEE and I see we get authenticated correctly but then lots of other info is spit out very quickly and I am having a hard time trying to understand it. I think it may have to do with the location table. But I could be wrong. Any ideas how to debug? Thank you.

2 3

Radius Generates Core Dump
by Omar Labs 188 Gmail 27 Jan '12

27 Jan '12

Hello: I was testing the Server Kamailio 3.2.1 with Accounting and Radius, it crash generating a Big Core Dump, and freezing up, If I remove the Radius module from configuration, accounting via the database works fine. Did anybody else have the same problem. Appears a Memory leak. Thanks

2 1

Kamailio for Wholesale Termination
by Abid Naseem 27 Jan '12

27 Jan '12

Dear *All*, I am new to voip world and interested in knowing if I we can use *KAMAILIO + * *SIREMIS *for wholesale voip termination? We buy direct routes and give it to resellers..currently using asterisk. appreciate your support! keep inspiring. -- Regards, *Abid Naseem* +44 (793) 741 5043 +1 (415) 562 6333

2 1

Kamailio with Private IP - ACK for 200 OK problem
by Krishna Kurapati 26 Jan '12

26 Jan '12

Hi, I am having trouble with setting Record Route when Kamailio is behind a Load balancer. The Load balancer acts as a static NAT where the public IP of the load balancer is mapped to private IP of the Kamailio. This is creating issues with ACK for 200 OK since Kamailio is stuffing the 200 OK with private-ip in the Record-Route. The UAC looks like trying to send ACK back to this private IP which is not reachable. Is there a configuration option to let kamailio use Public IP when setting record-route in 200 OK? Thanks for any help... Krish Kura

2 1

Memory usage problem in xmpp modules
by laura testi 26 Jan '12

26 Jan '12

Hi, we are using the XMPP gateway modules: • PUA • PUA_XMPP • XMPP The behavior is ok but since the modules are started, the memory usage is getting increased and now also the swap memory area is used. I send to you the following information: 1. Numer of records from PUA and xmpp_presentity tables, they are quite stable: SQL> select count(*) from pua; COUNT(*) ---------- 10218 SQL> select count(*) from xmpp_presentity; COUNT(*) ---------- 754 2. list of PKG fragments with gdb : $1 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $2 = {size = 32, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $3 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $4 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $5 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $6 = {size = 104, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $7 = {size = 32, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $8 = {size = 32, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $9 = {size = 56, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $10 = {size = 96, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $11 = {size = 56, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $12 = {size = 112, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $13 = {size = 24, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $14 = {size = 24, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $15 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $16 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $17 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160}…………. 3. Output of sercmd cfg.set_now_int core mem_dump_pkg : ……………….. 10969. N address=0xb1b320 frag=0xb1b2f0 size=104 used=1 qm_status: alloc'd from presence: utils_func.h: uandd_to_uri(66) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed qm_status: 10970. N address=0xb1b3e8 frag=0xb1b3b8 size=32 used=1 qm_status: alloc'd from presence: utils_func.h: uandd_to_uri(66) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed qm_status: 10971. N address=0xb1b468 frag=0xb1b438 size=32 used=1 qm_status: alloc'd from presence: utils_func.h: uandd_to_uri(66) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed qm_status: dumping free list stats : qm_status: hash= 1. fragments no.: 4, unused: 0 bucket size: 8 - 8 (first 8) qm_status: hash= 2. fragments no.: 8, unused: 0 bucket size: 16 - 16 (first 16) qm_status: hash= 3. fragments no.: 6, unused: 0 bucket size: 24 - 24 (first 24) qm_status: hash= 4. fragments no.: 1, unused: 0 bucket size: 32 - 32 (first 32) qm_status: hash= 6. fragments no.: 3, unused: 0 bucket size: 48 - 48 (first 48) qm_status: hash= 8. fragments no.: 8, unused: 0 bucket size: 64 - 64 (first 64) qm_status: hash= 9. fragments no.: 6, unused: 0 bucket size: 72 - 72 (first 72) qm_status: hash= 12. fragments no.: 5, unused: 0 bucket size: 96 - 96 (first 96) qm_status: hash= 14. fragments no.: 6, unused: 0 bucket size: 112 - 112 (first 112) qm_status: hash= 17. fragments no.: 32, unused: 0 bucket size: 136 - 136 (first 136) qm_status: hash= 19. fragments no.: 1, unused: 0 bucket size: 152 - 152 (first 152) qm_status: hash= 21. fragments no.: 1, unused: 0 bucket size: 168 - 168 (first 168) qm_status: hash= 29. fragments no.: 1, unused: 0 bucket size: 232 - 232 (first 232) qm_status: hash= 31. fragments no.: 2, unused: 0 bucket size: 248 - 248 (first 248) qm_status: hash= 33. fragments no.: 2, unused: 0 bucket size: 264 - 264 (first 264) qm_status: hash= 36. fragments no.: 1, unused: 0 bucket size: 288 - 288 (first 288) qm_status: hash= 57. fragments no.: 1, unused: 0 bucket size: 456 - 456 (first 456) qm_status: hash= 60. fragments no.: 1, unused: 0 bucket size: 480 - 480 (first 480) qm_status: hash= 76. fragments no.: 709, unused: 0 bucket size: 608 - 608 (first 608) qm_status: hash= 88. fragments no.: 1, unused: 0 bucket size: 704 - 704 (first 704) qm_status: hash= 116. fragments no.: 1, unused: 0 bucket size: 928 - 928 (first 928) qm_status: hash= 145. fragments no.: 1, unused: 0 bucket size: 1160 - 1160 (first 1160) qm_status: hash= 216. fragments no.: 2, unused: 0 bucket size: 1728 - 1728 (first 1728) qm_status: hash= 2049. fragments no.: 1, unused: 0 bucket size: 16384 - 32768 (first 22032) qm_status: hash= 2057. fragments no.: 1, unused: 0 bucket size: 4194304 - 8388608 (first 5982824) Is it a memory leak in one of these modules? Thanks in advance and kind regards, laura

5 6

get the kamailio up time and other params from the cfg file.
by Uri Shacked 26 Jan '12

26 Jan '12

Hi, is there a way to get the total up time of kamailio from the cfg file? like $stat(used_size) gives me the stat of the shared memory used size..... and if i am in it... is there a way to get the dispatcher status as well? i am trying to find a way to export these params with out the need to pull them using kamctl..... any ideas? thanks, Uri

4 3

Re: [SR-Users] detect the direction of the BYE
by Uri Shacked 26 Jan '12

26 Jan '12

On Thu, Jan 26, 2012 at 9:24 AM, Uri Shacked <uri.shacked(a)gmail.com> wrote: > Hi, > > i would like to detct the direction of the BYE. was it drom the caller or > the callee? > in the ACC module i can do that if i analyze the to and from tags... > how do i do it (in a nice way) in the cfg file? > > cheers. >

2 1

Call Control Behavior
by Reda Aouad 26 Jan '12

26 Jan '12

Hi, My question concerns Call Control. When a SIP call's timer ends, Call Control uses dlg_end_dlg MI command of the dialog module to send a BYE to the UA. Can I intercept the BYE message, drop it, change it and send instead an INVITE or a REDIRECT using (maybe) the UAC module? The INVITE/REDIRECT would connect the UA to a media gateway to announce a message when the session ends. Any ideas? This also leads to thoughts on accounting and rating. In the case of sending and INVITE, the rating of the call becomes inaccurate since there's no BYE when the call ends, but only after the media gateway's announcement ends. In the case of a REDIRECT, the UA sends a BYE to the established call before the INVITE to the media gateway, and thus rating remains totally accurate. What do you think? Thanks RA

1 0

Kamailio Redundancy examples/pointers required
by Sammy Govind 26 Jan '12

26 Jan '12

Hello list, I've been looking for ways to create a redundant Kamailio cluster. I've googled alot but haven't got any concrete or final wording on any one solution that'll just work perfectly. The basic requirement is that in case of Kamailio application failure or in case of physical machine error the whole setup just starts working on secondary server. Please suggest anything whichever you guys are using. Any reference URLs would be very much appreciated. Thanks and Regards, Sammy

2 2

Gentoo ebuild pkg v3.2.1
by caio 25 Jan '12

25 Jan '12

Hello, I want to announce the availability of an ebuild for version 3.2.1 on Gentoo Portage Overlays. Here the git link: http://goo.gl/JXLQ4 The unofficial overlays portage website: * http://gpo.zugaina.org/net-misc/sip-router Remember this is not an official ebuild, and neither it's not at official gentoo portage tree. Henning, can you add/update it at sip-router git repository (pkg/ser/gentoo)? Thank you and best regards, Claudio

1 0

access to the "Available Functions" in the modules & the way to bridge a call based on username and domain
by Xi Liu 25 Jan '12

25 Jan '12

Hi, I'm new to kamailio, have some questions here hope anyone can help! Question 1, I'm trying to use some of the functions in usrloc module, but for some reason if I use those from "Available Functions" part, the kamailio won't start and give an error of "unknown command <ul_get_ucontact>, missing loadmodule?" The usrloc module is definitely loaded, and it will start as long as I avoid using those "Available Functions". I tried some other modules it will lead to same error. Question 2, What is the simplest way to bridge a call based on username and domain? I was trying to make a simple experiment I used to do in Freeswitch, ========================== Have 1000,1001,1002 registered, When 1000 calls 9999, the calls will be bridged to 1001 or 1002 based on different time period. ========================== While in Freeswitch, it would be as simple as "bridge xxxx(a)domain.com", and then freeswitch will lookup contact information and forward the call. But in kamailio, I only succeed in forwarding calls by change the $du to the exact contact uri, which is "sip:xxxx@ip-of-the-phone:port", the call r_relay(). In order to do this, I need call a perl script to lookup the location database to find a match… I'm 99% sure I'm doing wrong here… there must be a way way easier than that, but so far I can't find it…. Please help… Thanks, Xi

2 2

UAC module + customize user_agent
by Douglas Adami 25 Jan '12

25 Jan '12

Hello everyone, I'm new here and I go to test my settings, but I need to authenticate a remote server, I was reading about the UAC module, one has configuration examples? I can also customize the user_agent authentication, because the service only allows specific user_agents? Thanks in advance! Regards, Doug

2 1

Re: [SR-Users] Radius and kamailio: avp unknown attribute
by DucDung 25 Jan '12

25 Jan '12

Hi man, I have the same problem like you posted in sip-router.org. The problem is that : rc_avpair_new: unknown attribute 5 I see that u have already resolved that problem before. So can u help me to fix it. Thank you very much. Please contact me follow these emails : ducdung0909(a)yahoo.com or ducdung0909(a)gmail.com .

2 1

Images missing
by sjs205.linux＠googlemail.com 25 Jan '12

25 Jan '12

Hello all, Just reading through some of the docs on the website; is it possible to get the images in this file: http://sip-router.org/docbook/sip-router/branch/master/sip/sip_introduction… An excellent introduction into SIP :D Kind Regards, Steven

2 1

Re: [SR-Users] Homer SIP Capture Setup Question
by JR Richardson 24 Jan '12

24 Jan '12

> On 01/16/2012 09:11 PM, JR Richardson wrote: > > > So my question; what is the difference between a drop; and exit; > > within the on_reply route > > Remember that an onreply_route is a callback that allows you to > intercept a reply, but the reply is forwarded by default afterward (if > using stateful forwarding). This is different from a request route, > where it's up to you to marshal everything, and to explicitly forward > the message. > > Thus, exiting an onreply_route has null effect. It doesn't do > anything to stop the reply from being forwarded, because forwarding is > the behaviour--the onreply_route is just a hook. In contrast, 'drop' > explicitly drops the message and prevents further forwarding of the > reply from taking place. > > This is different than in a request route, where exiting precludes any > forwarding action, because forwarding action must be taken explicitly > -- e.g. t_relay(). > > -- > Alex Balashov - Principal Aaaah, concise explanation, once again, thanks for turning on the light bulb Alex. JR

2 1

Is t_on_failure with uac_req_send possible?
by Nathan Angelacos 24 Jan '12

24 Jan '12

I'm trying to arm the failure route for a message sent via uac_req_send - similar to: { $uac_req(method)="OPTIONS"; $uac_req(ruri)="sip:kamailio.org"; $uac_req(furi)="sip:kamailio.org"; $uac_req(turi)="sip:kamailio.org"; t_on_failure("UAC_FAIL"); uac_req_send(); } failure_route("UAC_FAIL") { xlog ...... } If there is no response from the remote server, kamailio resends until the timers expire; but the failure route is not run. Similarly, if a 404 is sent back from the remote server, the failure route is not run. Am I missing something obvious? Thanks

3 3

(no subject)
by Florian Meister 24 Jan '12

24 Jan '12

unsubscribe

2 1

unable to find dialog for ACK with route param
by Uri Shacked 24 Jan '12

24 Jan '12

Hi, i use the dialog module. i create the dialog using dlg_manage() right after the invite is recieved. in a case where the final destination rejects the call (486 busy), i get the following error: " WARNING: dialog [dlg_handlers.c:1054]: unable to find dialog for ACK with route param....." i guess the dialog module deletes the dialog on the 486 reply and gives me this erro when an ACK for the 486 is recieved from the caller... no a bug issue, but, is there a way to overcome it? thanks, Uri

2 2

Problem with www_challenge
by Stoyan Mihaylov 23 Jan '12

23 Jan '12

Hi, If I am using IP address, I have no problems. If I use domain name (pointing to same IP address) then phones do not register. I have couple of hardware and software SIP phones and when I put name of domain (stribogkonsult.com) - then no registration, if I put IP - no problems. I put IP, and names in domain table. Effect is - when I use domain name, there are bulk of messages in syslog - like messages reenter kamailio again and again. xlog("Auth attemps $au\n"); - I receive many times if I use domain name, if I use IP - once only. Problem can be somewhere else of course. But everything else work as I expect. route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { if (!www_authorize("$td", "subscriber")) { xlog("RegisterChalange $au $td\n"); www_challenge("$td", "1"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } xlog("Auth attemps $au\n"); } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { xlog("Register MySelf\n"); if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; }

2 2

db_cassandra
by David 23 Jan '12

23 Jan '12

Using thrift/0.8.0/thrift-0.8.0.tar.gz I get this when trying to build db_cassandra. In file included from /usr/local/include/thrift/protocol/TProtocol.h:23, from /usr/local/include/thrift/protocol/TBinaryProtocol.h:23, from dbcassa_base.cpp:36: /usr/local/include/thrift/transport/TTransport.h:34: error: expected constructor, destructor, or type conversion before ‘readAll’ /usr/local/include/thrift/transport/TTransport.h:107: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TTransport.h:111: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TTransport.h:124: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TTransport.h:128: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TTransport.h:139: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TTransport.h:156: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TTransport.h:156: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TTransport.h:160: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TTransport.h:160: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TTransport.h:172: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TTransport.h:214: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TTransport.h:214: error: expected ‘;’ before ‘*’ token /usr/local/include/thrift/transport/TTransport.h:218: error: expected ‘;’ before ‘virtual’ /usr/local/include/thrift/transport/TTransport.h:218: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TTransport.h:218: error: ‘uint8_t’ declared as a ‘virtual’ field /usr/local/include/thrift/transport/TTransport.h:218: error: expected ‘;’ before ‘*’ token /usr/local/include/thrift/transport/TTransport.h:231: error: expected ‘;’ before ‘void’ /usr/local/include/thrift/transport/TTransport.h:231: error: ‘uint32_t’ has not been declared /usr/local/include/thrift/transport/TTransport.h:235: error: ‘uint32_t’ has not been declared /usr/local/include/thrift/transport/TTransport.h: In member function ‘void apache::thrift::transport::TTransport::write(int)’: /usr/local/include/thrift/transport/TTransport.h:158: error: ‘buf’ was not declared in this scope /usr/local/include/thrift/transport/TTransport.h:158: error: ‘len’ was not declared in this scope In file included from /usr/local/include/thrift/protocol/TBinaryProtocol.h:23, from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TProtocol.h: At global scope: /usr/local/include/thrift/protocol/TProtocol.h:184: error: expected constructor, destructor, or type conversion before ‘skip’ /usr/local/include/thrift/protocol/TProtocol.h:305: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:309: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:312: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:314: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:316: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:320: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:322: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:324: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:328: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:330: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:333: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:335: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:338: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:340: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:342: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:344: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:346: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:348: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:350: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:352: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:354: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:356: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:363: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:369: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:374: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:379: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:386: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:391: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:396: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:403: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:408: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:413: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:418: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:423: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:428: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:433: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:438: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:443: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:448: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:453: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:458: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:463: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:472: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:476: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:478: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:480: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:482: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:486: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:488: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:492: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:494: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:497: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:499: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:502: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:504: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:506: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:508: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:510: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:512: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:514: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:516: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:518: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:520: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:522: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:529: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:534: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:539: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:544: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:551: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:556: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:561: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:566: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:571: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:576: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:581: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:586: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:591: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:596: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:601: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:606: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:611: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:616: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:621: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:631: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:639: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:643: error: ‘uint32_t’ does not name a type In file included from /usr/local/include/thrift/protocol/TBinaryProtocol.h:24, from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TVirtualProtocol.h:42: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:52: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:57: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:63: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:68: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:78: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:83: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:91: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:96: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:103: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:108: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:115: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:120: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:126: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:132: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:138: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:144: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:150: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:156: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:162: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:168: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:174: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:184: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:190: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:196: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:201: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:211: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:216: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:221: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:231: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:236: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:243: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:248: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:255: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:260: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:266: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:272: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:278: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:284: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:290: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:296: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:302: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:308: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:329: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:336: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:341: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:345: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:349: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:356: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:360: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:364: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:370: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:374: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:379: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:383: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:388: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:392: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:396: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:400: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:404: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:408: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:412: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:416: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:420: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:428: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:435: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:439: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:443: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:447: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:454: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:458: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:464: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:468: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:473: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:477: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:482: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:486: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:490: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:494: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:498: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:502: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:506: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:510: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:514: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:518: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:522: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:536: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:548: error: ‘uint32_t’ does not name a type In file included from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TBinaryProtocol.h:92: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:96: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:99: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:101: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:103: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:107: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:109: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:111: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:115: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:117: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:119: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:121: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:123: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:125: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:127: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:129: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:131: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:133: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:135: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:137: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:139: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:146: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:150: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:152: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:154: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:156: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:160: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:162: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:166: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:168: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:170: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:172: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:174: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:176: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:180: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:182: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:184: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:186: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:188: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:190: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:192: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:195: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:208: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/protocol/TBinaryProtocol.h:208: error: expected ‘;’ before ‘*’ token /usr/local/include/thrift/protocol/TBinaryProtocol.h: In constructor ‘apache::thrift::protocol::TBinaryProtocolT<Transport_>::TBinaryProtocolT(boost::shared_ptr<X>)’: /usr/local/include/thrift/protocol/TBinaryProtocol.h:51: error: class ‘apache::thrift::protocol::TBinaryProtocolT<Transport_>’ does not have any field named ‘string_buf_’ /usr/local/include/thrift/protocol/TBinaryProtocol.h: In constructor ‘apache::thrift::protocol::TBinaryProtocolT<Transport_>::TBinaryProtocolT(boost::shared_ptr<X>, int32_t, int32_t, bool, bool)’: /usr/local/include/thrift/protocol/TBinaryProtocol.h:65: error: class ‘apache::thrift::protocol::TBinaryProtocolT<Transport_>’ does not have any field named ‘string_buf_’ /usr/local/include/thrift/protocol/TBinaryProtocol.h: In destructor ‘apache::thrift::protocol::TBinaryProtocolT<Transport_>::~TBinaryProtocolT()’: /usr/local/include/thrift/protocol/TBinaryProtocol.h:69: error: ‘string_buf_’ was not declared in this scope In file included from /usr/local/include/thrift/protocol/TBinaryProtocol.h:24, from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TVirtualProtocol.h: At global scope: /usr/local/include/thrift/protocol/TVirtualProtocol.h: In instantiation of ‘apache::thrift::protocol::TVirtualProtocol<apache::thrift::protocol::TBinaryProtocolT<apache::thrift::transport::TTransport>, apache::thrift::protocol::TProtocolDefaults>’: /usr/local/include/thrift/protocol/TBinaryProtocol.h:37: instantiated from ‘apache::thrift::protocol::TBinaryProtocolT<apache::thrift::transport::TTransport>’ /usr/local/include/thrift/protocol/TBinaryProtocol.h:259: instantiated from here /usr/local/include/thrift/protocol/TVirtualProtocol.h:554: error: no members matching ‘apache::thrift::protocol::TProtocolDefaults::readBool’ in ‘class apache::thrift::protocol::TProtocolDefaults’ In file included from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TBinaryProtocol.h: In instantiation of ‘apache::thrift::protocol::TBinaryProtocolT<apache::thrift::transport::TTransport>’: /usr/local/include/thrift/protocol/TBinaryProtocol.h:259: instantiated from here /usr/local/include/thrift/protocol/TBinaryProtocol.h:178: error: no members matching ‘apache::thrift::protocol::TVirtualProtocol<apache::thrift::protocol::TBinaryProtocolT<apache::thrift::transport::TTransport>, apache::thrift::protocol::TProtocolDefaults>::readBool’ in ‘class apache::thrift::protocol::TVirtualProtocol<apache::thrift::protocol::TBinaryProtocolT<apache::thrift::transport::TTransport>, apache::thrift::protocol::TProtocolDefaults>’ In file included from /usr/local/include/thrift/protocol/TBinaryProtocol.h:277, from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:31: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:51: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:56: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:62: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:67: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:78: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:83: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:89: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:100: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:105: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:114: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:119: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:128: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:133: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:140: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:146: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:153: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:160: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:167: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:179: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:189: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:198: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ In file included from /usr/local/include/thrift/protocol/TBinaryProtocol.h:277, from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:230: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:235: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:241: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:246: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:263: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:268: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:289: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:294: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:312: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:317: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:335: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:340: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:348: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:356: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:367: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:378: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:389: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:404: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:412: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:417: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ In file included from /usr/local/include/thrift/transport/TSocket.h:33, from dbcassa_base.cpp:37: /usr/local/include/thrift/transport/TVirtualTransport.h:50: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TVirtualTransport.h:53: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TVirtualTransport.h:56: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TVirtualTransport.h:56: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TVirtualTransport.h:59: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TVirtualTransport.h:59: error: expected ‘;’ before ‘*’ token /usr/local/include/thrift/transport/TVirtualTransport.h:62: error: expected ‘;’ before ‘void’ /usr/local/include/thrift/transport/TVirtualTransport.h:62: error: ‘uint32_t’ has not been declared /usr/local/include/thrift/transport/TVirtualTransport.h: In member function ‘void apache::thrift::transport::TTransportDefaults::write(int)’: /usr/local/include/thrift/transport/TVirtualTransport.h:57: error: ‘buf’ was not declared in this scope /usr/local/include/thrift/transport/TVirtualTransport.h:57: error: ‘len’ was not declared in this scope /usr/local/include/thrift/transport/TVirtualTransport.h: At global scope: /usr/local/include/thrift/transport/TVirtualTransport.h:94: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TVirtualTransport.h:98: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TVirtualTransport.h:102: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TVirtualTransport.h:102: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TVirtualTransport.h:106: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TVirtualTransport.h:106: error: ‘uint8_t’ declared as a ‘virtual’ field /usr/local/include/thrift/transport/TVirtualTransport.h:106: error: expected ‘;’ before ‘*’ token /usr/local/include/thrift/transport/TVirtualTransport.h:110: error: expected ‘;’ before ‘virtual’ /usr/local/include/thrift/transport/TVirtualTransport.h:110: error: ‘uint32_t’ has not been declared /usr/local/include/thrift/transport/TVirtualTransport.h:124: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TVirtualTransport.h: In member function ‘virtual void apache::thrift::transport::TVirtualTransport<Transport_, Super_>::write_virt(int)’: /usr/local/include/thrift/transport/TVirtualTransport.h:103: error: ‘buf’ was not declared in this scope /usr/local/include/thrift/transport/TVirtualTransport.h:103: error: ‘len’ was not declared in this scope In file included from dbcassa_base.cpp:37: /usr/local/include/thrift/transport/TSocket.h: At global scope: /usr/local/include/thrift/transport/TSocket.h:100: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TSocket.h:105: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TSocket.h:105: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TSocket.h:110: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TSocket.h:216: error: ISO C++ forbids declaration of ‘sockaddr’ with no type /usr/local/include/thrift/transport/TSocket.h:216: error: expected ‘;’ before ‘*’ token /usr/local/include/thrift/transport/TSocket.h:237: error: ISO C++ forbids declaration of ‘sockaddr’ with no type /usr/local/include/thrift/transport/TSocket.h:237: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TSocket.h:290: error: ‘sockaddr_in’ does not name a type /usr/local/include/thrift/transport/TSocket.h:291: error: ‘sockaddr_in6’ does not name a type In file included from /usr/local/include/thrift/transport/TTransportUtils.h:29, from dbcassa_base.cpp:38: /usr/local/include/thrift/transport/TBufferTransports.h:62: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TBufferTransports.h:75: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TBufferTransports.h:94: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TBufferTransports.h:94: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TBufferTransports.h:107: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TBufferTransports.h:107: error: expected ‘;’ before ‘*’ token dbcassa_base.cpp:1031: error: expected ‘;’ at end of input dbcassa_base.cpp:1031: error: expected ‘}’ at end of input In file included from /usr/local/include/thrift/transport/TTransportUtils.h:29, from dbcassa_base.cpp:38: /usr/local/include/thrift/transport/TBufferTransports.h: In member function ‘void apache::thrift::transport::TBufferBase::write(int)’: /usr/local/include/thrift/transport/TBufferTransports.h:95: error: ‘new_wBase’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h:95: error: ‘wBase_’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h:95: error: ‘len’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h:96: error: ‘wBound_’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h:97: error: ‘buf’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h:101: error: ‘buf’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h:101: error: ‘writeSlow’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h: At global scope: /usr/local/include/thrift/transport/TBufferTransports.h:102: error: expected unqualified-id at end of input /usr/local/include/thrift/transport/TBufferTransports.h:102: error: expected ‘}’ at end of input /usr/local/include/thrift/transport/TBufferTransports.h:102: error: expected ‘}’ at end of input /usr/local/include/thrift/transport/TBufferTransports.h:102: error: expected ‘}’ at end of input make[1]: *** [dbcassa_base.o] Error 1 make: *** [install-modules] Error 1

3 10

set acc flag to dlg_bye
by Uri Shacked 23 Jan '12

23 Jan '12

thanks..... i already did. no bye is inserted to the DB. 2 invites for the 183 and 200, 1 ack, and that is it... another idea? :-) " Hi, set acc flag in local_route, On Sat, Jan 21, 2012 at 1:08 PM, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> wrote: " >* Hi, *>* *>* i wonder, anyone has a good idea of how to set a flag, that will insert *>* the bye kamailio created to end the dialog into the acc table? *>* i am checking "early media", "cancel", "ack" etc.... but when kamaiilo *>* creates the bye with the dlg_bye command, there is no acc raw for that.... *>* i can create a sqlops and insert with the call's params... but there must *>* be a nicer way, no? *>* *>* BR, *>* Uri *>* *

2 4

acc flag before droping the call
by Uri Shacked 22 Jan '12

22 Jan '12

Hi, I am doing an internal chaek right after the following: " ........... # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting ............. " sometimes this check does not find the conditions needed and creates a sl_reply (404). the scenario works OK. now, i need the acc tabale record.... should probably be "INVITE".....""....... ""..... "404" .... "not found".... well, there is no record there. i tried the ACCFAILED flag as well, still does not work. i guess it has something to do with the transaction stage.... so, any idea on how to get ACC record with the ACC module there? BR, Uri

2 1

set acc flag to dlg_bye
by Uri Shacked 21 Jan '12

21 Jan '12

Hi, i wonder, anyone has a good idea of how to set a flag, that will insert the bye kamailio created to end the dialog into the acc table? i am checking "early media", "cancel", "ack" etc.... but when kamaiilo creates the bye with the dlg_bye command, there is no acc raw for that.... i can create a sqlops and insert with the call's params... but there must be a nicer way, no? BR, Uri

2 1

SIP Service not Connecting to VoIP Provider
by IPAN Gov-Europe 20 Jan '12

20 Jan '12

Sirs: Our SIP service with you (pbx(a)iptel.org) is not connecting to the VoIP Provider we installed. It was working until today... What can we do to fix this? Frank Riley, CTO IPANGov www.IPANGov.edu.tc

3 2

onreply _route syntax error for homer capture server
by JR Richardson 20 Jan '12

20 Jan '12

Hi All, Tuning my homer capture server and ran into a syntax error in the onreply_route. The idea is to NOT to capture NOTIFY|SUBSCRIBE|OPTIONS. The route block compiles ok but I get the following errors for the lines in the onreply_route: 0(6443) : <core> [cfg.y:3504]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 48, column 46: syntax error 0(6443) : <core> [cfg.y:3504]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 49, column 15: bad expression 0(6443) : <core> [cfg.y:3504]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 49, column 16: ERROR: bad config file (3 errors) route { if (!(method =~ "^(NOTIFY|SUBSCRIBE|OPTIONS)")) { sip_capture(); } exit; } 47 onreply_route { 48 if (!($rm =~ "^(NOTIFY|SUBSCRIBE|OPTIONS)") { 49 sip_capture(); 50 } 51 drop; 52 } This example was used from the homer wiki http://code.google.com/p/homer/wiki/Kamailio root@homer:/etc/kamailio# kamailio -V version: kamailio 3.2.1 (x86_64/linux) bee094 Any guidance is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

2 2

Group-based routing (Carrierroute - DRouting)
by Carlo Dimaggio 20 Jan '12

20 Jan '12

Hi all, I need group-based routing, that is I want a routing table for each type of subscriber (residential, business,...). I have seen that Drouting does it throught groupID and I'm wondering about the same behaviour with carrierroute module (as LCR uses only flat costs). Can the "carrier" parameter be the right way to do it? I have no special requirement about the number of routes. However, Is the DRouting actively maintained? Regards,

3 2

I am unable to call other iptel users and my voice mail is not working
by praseeth k m 20 Jan '12

20 Jan '12

Hi Team, I am unable to call other iptel users and my voice mail is not working . Would you be able to help me on this ? my user id is praseeth.panicker Regards, Praseeth

2 1

ip address error with exec module
by Uri Shacked 19 Jan '12

19 Jan '12

Daniel hi, a little late but, here is the level 3 debug log.... Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: tm [t_reply.c:1509]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [receive.c:291]: receive_msg: cleaning up Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [receive.c:291]: receive_msg: cleaning up Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=860 a=65 n=assign Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=861 a=26 n=exec_msg Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: : exec [../../parser/../ip_addr.h:688]: BUG: ip_addr2sbuf: unknown address family 0 Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: <core> [md5utils.c:67]: DEBUG: MD5 calculated: 64d3de5c2447acd6006b8539d1246e10 Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_LENGTH=0 Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_hf.c:267]: SIP_HF_CSEQ=1 OPTIONS Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_hf.c:267]: SIP_HF_CALLID=123 Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_hf.c:267]: SIP_HF_TO=<you(a)kamailio.org> Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_hf.c:267]: SIP_HF_FROM=<you(a)kamailio.org>;tag=123 Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_hf.c:267]: SIP_HF_VIA=SIP/2.0/UDP 127.0.0.1 Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_mod.c:175]: executing [nc -u -w0 10.2.0.54 514 <<< ",denied_calls-0,dos-green"] Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: <core> [main.c:852]: SIGCHLD received: we do not worry about grand-children Jan 18 07:36:25 kamailiolabroute /usr/local/sbin/kamailio[2390]: DEBUG: tm [t_reply.c:1168]: ->>>>>>>>> T_code=0, new_code=408 Jan 18 07:36:25 kamailiolabroute /usr/local/sbin/kamailio[2390]: DEBUG: tm [t_reply.c:1931]: DEBUG: local_reply: branch=0, save=0, winner=0 Jan 18 07:36:25 kamailiolabroute /usr/local/sbin/kamailio[2390]: DEBUG: tm [t_reply.c:1968]: DEBUG: local transaction completed Jan 18 07:36:25 kamailiolabroute /usr/local/sbin/kamailio[2390]: DEBUG: tm [t_hooks.c:288]: DBG: trans=0xb00a772c, callback type 256, id 0 entered Jan 18 07:36:25 kamailiolabroute /usr/local/sbin/kamailio[2390]: DEBUG: dispatcher [dispatch.c:2345]: OPTIONS-Request was finished with code 408 (to sip:10.10.21.26, group 47)

2 3

Kamailio Keepalive
by Reda Aouad 18 Jan '12

18 Jan '12

Hi, I am using the nat_keepalive function of the nat_traversal module, using the OPTIONS message, and my code is the following : if (is_method("REGISTER")) nat_keepalive(); I see that Kamailio still sends OPTIONS packets even if the client unregisters itself. It stops later, but I don't know when exactly. Is that a normal behaviour? What is the keepalive timeout? Why doesn't Kamailio remove the client from its keepalive list if it unregisters and is no more in the user location? And is there a way to remove a contact from the user location if it doesn't respond to keepalive packets? Thanks Reda

2 1

re-route SIP response message
by Robert R 18 Jan '12

18 Jan '12

Hi, Can I re-route a SIP response message? In other words, after the SIP proxy receives a 503 from node A, I need to change the TO field and re-route it to node B. I tried to modify $rd or $rU and didn't work. Thanks, R

2 1

New feature: embedded MSRP relay
by Daniel-Constantin Mierla 18 Jan '12

18 Jan '12

Hello, a new module is now on development branch, named msrp, offering an embedded MSRP relay. Message Session Relay Protocol is specified by IETF ( core and relay part in RFC4975 and RFC4976), one of typical use cases is Instant Messaging sessions negotiated via INVITE-200ok-ACK, a different approach than SIP MESSAGE request. There is no external dependency, the transport layer from core is reused to manage TCP/TLS connections used for MSRP. A MSRP relay is pretty much an independent node, thus you can run Kamailio just to deal with MSRP traffic. But there is no problem to run Kamailio to handle SIP and MSRP traffic in the same instance, same or different sockets (it is working even when SIP/MSRP are sent over the same connection or different connections to same port). I couldn't find a reliable and trustable open source SIP phone to test with it so far (hints are welcome), the module was tested with network tools and chaining Kamailio instances. Help with testing and feedback is very appreciated -- I can provide guidelines to adjust the config file to fit tester's needs, just contact me off list or via IRC channel. For the moment, the relation (user,session) management is done in the config, using htable module for example (see the README), it may be added inside the module for the future to make it easier overall, although is less than 10 lines of config -- this is mainly for extra security reasons, to check if the session id matches the connection that was opened when the session was created, otherwise the user authentication functionality is done reusing existing functions from auth module. Another benefit is performances, storing local socket and connection IP/port saves some time to lookup the connection. Building MSRP relay on top of SIP server was done first for the benefit of reusing the transport layer from the core for IPv4/IPv6 and TCP/TLS, which is mature, scalable and offers asynchronous communication. Besides that, you have most of the config tools to route SIP requests available for routing MSRP (authentication, authorization, IP checking, accounting, a.s.o.). I made a news post with more details: * http://www.kamailio.org/w/2012/01/new-module-embedded-msrp-relay/ The readme of the new module is available at: * http://kamailio.org/docs/modules/devel/modules/msrp.html Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 6

sip-router and related projects at FOSDEM 2012 - social event
by Stefan Sayer 17 Jan '12

17 Jan '12

Hello everyone, as has become already a tradition, we would like to invite all VoIP people and everyone interested in sip-router and related projects (SER, Kamailio/OpenSER, OpenSIPS, SEMS etc) who is attending FOSDEM, Europe's biggest Free and Open Source conference, to a dinner on the evening of Saturday February 4th. This is the perfect opportunity to socialize among the developers, get to know the people behind email addresses and commits, do networking, find the people for the next project, and/or just hang out together with fine food and drinks. Participation at the event is free of charge, everybody pays for their own expenses. In order to do reservation in a restaurant please send me a short mail if you would like to attend. Please also indicate if you don't want to have your contact details communicated to the other social event attendees. If you are still wondering whether it's worth to go to FOSDEM - among all the other interesting talks this year in the telephony-devroom on Sunday, 5th, there will be a few directly related talks (the exact time may be subject to change): 11:30 Andreas Granig: From zero to VoIP provider in 15 minutes 12:55 Stefan Sayer: Session Border Control with SEMS 15:25 Daniel-Constantin Mierla: Secure SIP Communications with Kamailio 16:15 Saúl Ibarra Corretgé: SIP beyond VoIP http://fosdem.org/2012/schedule/track/telephony_and_communications_devroom Hope to see you all in Brussels! With Best Regards Stefan Sayer -- tel:+491621366449 sip:sayer@iptel.org mailto/xmpp:stefan.sayer@gmail.com

3 2

new developer: Anca Vamanu
by Henning Westerholt 17 Jan '12

17 Jan '12

Hello all, I'm happy to announce a new developer for the project: Anca Vamanu. Welcome to the development team! Anca joined 1&1 last year and worked since then on several internal modules and also many important bug fixes for our kamailio server setup. She is a experienced C and C++ developer and contributed in the past a lot to OpenSER and other VoIP projects. One interesting work she just finished and that we also like to contribute to the community is a db_cassandra module, providing access to this NoSQL database [1] from kamailio. She will take over the maintenance of this module and also work on improving other parts of the server. Best regards, Henning [1] http://cassandra.apache.org/ -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany

3 2

Homer SIP Capture Setup Question
by JR Richardson 17 Jan '12

17 Jan '12

Hi All, I spent a few hours this afternoon setting up a Homer SIP Capture server today. The build went well, good example setup howto's. My setup is a single kamailio/sipcature/homer node sitting on a LAN with 2 NIC's, eth0 for network access ssh/web, eth1 is the capture port connected to switch mirrored port. On the switch I have a kamailio proxy server being monitored. All SIP traffic through the proxy server is logged to the homer server, but the homer server is receiving SIP packets on the capture interface and sending out SIP messages through the network access eth0 port. The messages are being processed from the on_reply route. The default config for the capture node is to capture all SIP packets, but this seems to create the problem. Here is the route and onreply_route default config: route { sip_capture(); exit; } onreply_route { sip_capture(); exit; } If I replace the onreply_route exit; with drop; then the capture server no longer sends SIP messages out the eth0 interface and everything seems to work as expected. So my question; what is the difference between a drop; and exit; within the on_reply route and how the hell is the capture server sending messages out the eth0 network interface when the server is 100% bound to eth1 capture interface? Any insight or guidance is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

NoSQL Backends for Subscriber Database.
by David 16 Jan '12

16 Jan '12

I was wondering if any NoSQL adapters are available for Kamailio to do "subscriber authentication" backend? I was looking at Mongo or CouchDB as two possible backends Seems like these would be good candidates for Backends for Kamailio. Any ideas on this? Thanks David.

2 1

Selecting a database for Kamailio, PostgreSQL or MySQL
by Spencer Thomason 16 Jan '12

16 Jan '12

Hello all, We are currently in the process of reworking our billing platform which uses Kamailio for routing in conjunction with a Django based web interface. I am curious if anyone has any feedback on which database engine they chose for their deployment and why. Obviously performance is a concern but data integrity is essential. Currently we have two Kamailio based clusters in two datacenters using MySQL in a dual master setup for each of the primary database nodes. I've been looking into PostgreSQL as a possible alternative and would like to hear any community thoughts. Thanks, Spencer

2 1

Re: [SR-Users] sr-users Digest, Vol 80, Issue 49
by Javier Gallart 16 Jan '12

16 Jan '12

Hello Yaron can you try "mi mt_reload"? Regards Javi > > Message: 2 > Date: Mon, 16 Jan 2012 11:54:04 +0200 > From: Yaron Nachum <nachum.yaron(a)gmail.com> > Subject: [SR-Users] using XMLRPC module for fifo commands > To: sr-users(a)lists.sip-router.org > Message-ID: > <CADKX-JDqsxBb3T6XOsiUUs=Xj+idNAXEECBx1HeuAu8dSEdbsA(a)mail.gmail.com > > > Content-Type: text/plain; charset="iso-8859-1" > > Hi everyone, > Is it possible to use the xmlrpc module in order to send fifo commands to > the kamailio. for example I would like to use it in order to trigger mtree > table reload. > > I have already enabled the xmlrpc module and it responds. however I didn't > succeeded in sending the appropriate syntax in lrder to make the mtree > reload work. > > I tried using mtree.mt_reload method or fifo.mt_reload . > > Does anyone have a clue? > > Yaron. >

2 2

using XMLRPC module for fifo commands
by Yaron Nachum 16 Jan '12

16 Jan '12

Hi everyone, Is it possible to use the xmlrpc module in order to send fifo commands to the kamailio. for example I would like to use it in order to trigger mtree table reload. I have already enabled the xmlrpc module and it responds. however I didn't succeeded in sending the appropriate syntax in lrder to make the mtree reload work. I tried using mtree.mt_reload method or fifo.mt_reload . Does anyone have a clue? Yaron.

1 0

drouting - do_routing with $avp does not work
by Uri Shacked 16 Jan '12

16 Jan '12

Hi, this is my cfg: if(do_routing("0")){ $avp(tdr)=$avp(dr_attrs); } else { $avp(tdr)="1"; } subst_user('/(.*)/$avp(Ssrvindx)/'); xlog("L_CRIT","$C(rg) group is $avp(tdr)$C(xx)\n"); if(!do_routing("$avp(tdr)")){ xlog("L_CRIT","$C(rg)No TDR is found for $avp(DID) with Index $avp(Ssrvindx)$C(xx)\n"); t_reply("404", "Not found"); exit; } in the log, i get the following: 1(25799) CRITICAL: <script>: group is 8 1(25799) ERROR: drouting [drouting.c:706]: failed to get group id when i change the do_routing("$avp(tdr)") to do_routing("8"), everything works..... it looks like with the avp (although it is 8) it does not work. any ideas?

2 1

Preventing the use of received port as the source port for next hop
by Arif Tuhin 16 Jan '12

16 Jan '12

I've noticed this in both kamailio 3.1.5 and 3.2.x. I've configured to receive on udp 9090 port. This sip message will be forwarded to a asterisk box listening on 5060 port. Now what i'm trying to achieve is , When kamailio will receive the sip message on udp 9090 port it will forward the message to asterisk using source port 5060, not 9090. How can i configure this?Thanks in advance. With Best Regards Ariful Hossain Tuhin email: 1. etothepowerpi(a)gmail.com 2. etothepowerpi(a)hotmail.com 3.etothepowerpi(a)yahoo.com skype: freeburn1986

2 1

when does ds_select_dst update list of addresses to choose from when using DB?
by Albert Petit 13 Jan '12

13 Jan '12

I'm using dispatcher module for load balancnig purposes I read active/inactive nodes from the database So I define on my kamailio.cfg: modparam("dispatcher", "db_url", "mysql://openser:openserrw@ <DB_IP>/openser") Then, on my route(DISPATCH) I call: * ds_select_dst("1", "4");* * t_relay();* * * My question is: Which *set of addresses is using **ds_select_dst *to choose destination. Will it do a query on the db each time updating active and inactive nodes on each ds_select_dst call? If not, when is the list of addresses to choose updated? (on kamaiilio restart, periodically...) I say this because I would like to update active and inactive nodes from a 3rd party application modifying the flags column but I need to know also how many time will it take for the change to be applied. BR Albert -- Albert

2 2

new feature: per-socket number of workers
by Daniel-Constantin Mierla 13 Jan '12

13 Jan '12

Hello, the master branch has a new feature that allow setting number of worker processes to handle received traffic per listen socket. So far there were global parameters that were applied to all sockets (e.g., 'children' value set the number of workers for all udp sockets). So far each UDP and SCTP socket had its own pool of workers (e.g., children=4 and 2 udp sockets resulted in 8 processes), while for tcp and tls was a single pool of workers (e.g., having children (or tcp_children) set 8, resulted in 8 processes no matter how many tcp/tls sockets). The new features is based on using a new config parameter, named "socket_workers", before a "listen" parameter. The value of socket_workers will overwrite the value of *children parameter. For UDP and SCTP will result in creating a number of "socket_workers" processes. For TCP and TLS will add an extra set of "socket_workers" processes, that will handle traffic only on those specific sockets. The value of "socket_workers" is reset with the next listen socket added. If "socket_workers" is not set, the value of "*children" parameter is used in backward compatible fashion. Some typical scenarios where this feature may become handy: - set a lower number for loopback or internal/replication sockets, as the traffic there is low (e.g, maybe for keepalive monitoring on loopback, or it is only REGISTER requests replication done over the replication sockets) - set a dedicated group of tcp/tls workers for handling http/xmlrpc/xcap traffic - handling such traffic may be time consuming, in this way you avoid delays on routing SIP over tcp/tls - fine tune the number of over all forked processes by a SIP server instance, thus controlling better the resources used from the physical server (e.g., overall private memory used by sip server is a matter of how many forked processes are there) You can see details about the new parameter and examples on the wiki page: * http://www.kamailio.org/wiki/cookbooks/devel/core?&#socket_workers Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

ip address error with exec module
by Uri Shacked 12 Jan '12

12 Jan '12

Hi, i am doing this command in a timer under kamailio: $var(log)='"'+'denied_calls-'+$stat(denied_calls)+',dos-'+$shv(dos)+'"'; exec_msg("nc -u -w0 10.2.0.54 514 <<< $var(log)"); each time the command is executed, i get the following error: 9(19302) : exec [../../parser/../ip_addr.h:688]: BUG: ip_addr2sbuf: unknown address family 0 9(19302) ERROR: exec [exec.c:86]: cmd nc -u -w0 10.2.0.54 514 <<< ",denied_calls-0,dos-green" failed. exit_status=-1, errno=32: Broken pipe any idea? the syslog server gets the message.... thanks, Uri

2 8

send bye to caller after dialog timeout
by Uri Shacked 12 Jan '12

12 Jan '12

hi, i am manging dialogs in memory only. no DB. i would like to set a timeout for a dialog and when the timeout is reached, to send "bye" to the parties. today i know how to timeout the dialog, and using the dlg_bye function is understood. but, how do i activate the dlg_bye when the time out is executed and whith which parameters in order to disconnect the right session? the explenation about the "route" to perform is not clear? please, thanks, Uri

2 3

"out of memory" in Kamailio 3.2
by Henry Fernandes 11 Jan '12

11 Jan '12

I upgraded from Kamailio 3.1 to Kamailio 3.2 (x86_64/linux) 31c991 recently. Since upgrading, my LCR does not work. The "load_gws" command always returns a 2 (meaning no routes found). When I do a "sercmd lcr.dump_rules", I get the following error: # sercmd lcr.dump_rules error: 500 - Internal server error: out of memory I've tried increasing "binrpc_max_body_size" and "binrpc_struct_max_body_size", but this doesn't help. -H

3 6

Hashed Passwords
by Ali Jawad 11 Jan '12

11 Jan '12

Hi All We are using plain text passwords for authentication, for the obvious reasons we want to change to hashed passwords so first I executed the following on the SQL server : update subscriber set ha1 = md5(concat(username, ':', domain, ':', password)), ha1b = md5(concat(username, '@', domain, ':', domain, ':', password)) The related config is : # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "ha1") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) and # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "ha1") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", 1) But I keep getting unauthorized, what did I miss here. Regards

3 4

Re: [SR-Users] RTPproxy on Kamailio 3.2.1 difficulty.
by Sammy Govind 10 Jan '12

10 Jan '12

Congrats so now RTPproxy issue has been resolved and Kamailio is saying that support for it has been enables. Next thing is ...ignore the reverse resolving of IPv6 IP !! thats not an issue at all for now. The error you are receiving is due to sip PUBLISH method ... that can be a bug in Kamailio or anything on which I can't comment any better than you. I hope some other gy take over the thread from here and see if this _IS_ really an issue or not, if so why is this here then !! Regards, Sammy. On Fri, Jan 6, 2012 at 3:37 PM, nunu abe <nunu_abe(a)yahoo.com> wrote: > Hi, > > I commented out the unix socket in the modparam as instructed and there is > still no change. > > 1. The rtpproxy was listening to port 7722, but to make sure I stopped it > and restarted it with the command: > xxx@DualStackCS:~$ sudo netstat -pln |grep rtpp > udp 0 0 127.0.0.1:7722 0.0.0.0:* > 860/rtpproxy > > xxx@DualStackCS:~$ sudo /etc/init.d/rtpproxy restart > Restarting RTP relay: rtpproxy. > > xxxxx@DualStackCS:~$ sudo netstat -pln |grep rtpp > udp 0 0 127.0.0.1:7722 0.0.0.0:* > 2303/rtpproxy > > xxxx@DualStackCS:~$ sudo /etc/init.d/rtpproxy stop > Stopping RTP relay: rtpproxy. > > xxxxx@DualStackCS:~$ sudo netstat -pln |grep rtpp > xxxxx@DualStackCS:~$ sudo rtpproxy -F -l 10.10.10.10 -6 > /3001:0:0:1::10 -s udp:127.0.0.1:7722 > xxxxx@DualStackCS:~$ sudo netstat -pln |grep rtpp > udp 0 0 127.0.0.1:7722 0.0.0.0:* > 2366/rtpproxy > > 2.Kamailio service started without error message but with this warning: > 0(1549) WARNING: <core> [socket_info.c:1275]: WARNING: fix_hostname: > could not rev. resolve 3001:0:0:1:0:0:0:10 > > 16(2422) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child > (11): nathelper > 16(2422) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child > (11): rtpproxy > 15(2421) INFO: rtpproxy [rtpproxy.c:1415]: rtp proxy > <udp:localhost:7722> found, support for it enabled > > 3.As soon as clients register, these errors appear in the logs > (similar errors appear even more frequently while making a call, shown in > the attachment): > 7(2407) DEBUG: <core> [parser/msg_parser.c:630]: method: <PUBLISH> > 7(2407) DEBUG: <core> [parser/msg_parser.c:632]: uri: <sip:300@ > [3001:0:0:1::10]> > 7(2407) DEBUG: <core> [parser/msg_parser.c:634]: version: <SIP/2.0> > 7(2407) DEBUG: <core> [parser/parse_via.c:1288]: Found param type 232, > <branch> = <z9hG4bK3847>; state=16 > 7(2407) DEBUG: <core> [parser/parse_via.c:2563]: end of header reached, > state=5 > 7(2407) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via > found, flags=2 > 7(2407) DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is > the first via > 7(2407) DEBUG: <core> [receive.c:147]: After parse_msg... > 7(2407) DEBUG: <core> [receive.c:188]: preparing to run routing scripts... > 7(2407) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=464 a=6 n=route > 7(2407) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=573 a=17 n=if > 7(2407) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=568 a=26 n=mf_process_maxfwd_header > > 6(2406) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=744 a=26 n=is_method > 6(2406) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=767 a=25 n=consume_credentials > 6(2406) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=780 a=3 n=return > 6(2406) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=489 a=26 n=remove_hf > 6(2406) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=500 a=6 n=route > 6(2406) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=704 a=27 n=sl_send_reply > > 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=467 a=6 n=route > 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=789 a=17 n=if > 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=789 a=57 n=force_rport > 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=799 a=17 n=if > 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=790 a=26 n=nat_uac_test > 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=795 a=17 n=if > 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=791 a=26 n=is_method > 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=792 a=25 n=fix_nated_register > 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=796 a=40 n=setflag > 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=799 a=3 n=return > 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=470 a=6 n=route > 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=617 a=17 n=if > 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] > l=582 a=25 n=has_totag > > So still lost in this issue. > Thanks for your help! > > Regards, > Maedot > ------------------------------ > *From:* Sammy Govind <govoiper(a)gmail.com> > *To:* nunu abe <nunu_abe(a)yahoo.com>; SIP Router - Kamailio (OpenSER) and > SIP Express Router (SER) - Users Mailing List < > sr-users(a)lists.sip-router.org> > *Sent:* Friday, January 6, 2012 10:06 AM > > *Subject:* Re: [SR-Users] RTPproxy on Kamailio 3.2.1 difficulty. > > Don't forget to change the modparams for RTPproxy as > > modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722") > # modparam("rtpproxy", "rtpproxy_sock", > "unix:/var/run/rtpproxy/rtpproxy.sock") > > > On Fri, Jan 6, 2012 at 2:05 PM, Sammy Govind <govoiper(a)gmail.com> wrote: > > Hi, > > Kamailio is trying to communicate to RTproxy using port 7722 , execute > the command > > # netstat -pln | grep rtpp > > and see if your rtpproxy is even listening on that interface, which it > isn't for sure. > > No if above is true then stop rtpproxy and start rtpproxy using the b/m > command > > #sudo rtpproxy -F -l 10.10.10.10 -6 /3001:0:0:1::10 -s udp:127.0.0.1:7722 > > confirm using the netstat command that RTPproxy is indeed listening on > 7722 port, if yes restart Kamailio and make sure you don't see kamailio > complaining about not connecting to RTPproxy anymore. > > > Regards, > Sammy. > > On Fri, Jan 6, 2012 at 1:31 PM, nunu abe <nunu_abe(a)yahoo.com> wrote: > > Hi, > > I have already posted this message but it was sent only to Sammy Govind > somehow, not to the mailing list. So I am posting it again to the list. I > am sorry if this is double posting or if this is not the right way to do it. > > **Original message*** > > > Hi Sammy > These are the params from my kamilio cfg file. Just a reminder, the > whole cfg file is copied from the link: > http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6, i just edited the > IP addresses and DB_URL line to fit my settings. > #!ifdef WITH_NAT > # ----- rtpproxy params ----- > # modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722") > modparam("rtpproxy", "rtpproxy_sock", > "unix:/var/run/rtpproxy/rtpproxy.sock") > > # ----- nathelper params ----- > modparam("nathelper", "natping_interval", 30) > modparam("nathelper", "ping_nated_only", 1) > modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) > modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") > > # params needed for NAT traversal in other modules > modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") > modparam("usrloc", "nat_bflag", FLB_NATB) > #!endif > I have tried to use both unix and UDP sockets each at a time.I removed the > square braces from the IP addresses in the rtpproxy default files and its > no more complainig about the "rtpproxy: setbindhost: No addressassociated with hostname" > but still have the same messages in the logs. > Thank you four your help. > Regards, > Maedot > > *From:* Sammy Govind <govoiper(a)gmail.com> > > *To:* nunu abe <nunu_abe(a)yahoo.com>; SIP Router - Kamailio (OpenSER) and > SIP Express Router (SER) - Users Mailing List < > sr-users(a)lists.sip-router.org> > *Sent:* Thursday, January 5, 2012 8:13 PM > > *Subject:* Re: [SR-Users] RTPproxy on Kamailio 3.2.1 difficulty. > > hi, > > quick look at your logs, > > 14(1596) ERROR: rtpproxy [rtpproxy.c:1466]: can't connect to RTP proxy > 15(1597) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (10): nathelper > 14(1596) ERROR: rtpproxy [rtpproxy.c:1538]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it > 5(1584) DEBUG: db_mysql [km_my_con.c:123]: server version is 5.1.41-3ubuntu12.10 > 14(1596) WARNING: rtpproxy [rtpproxy.c:1395]: can't get version of the RTP proxy > 14(1596) WARNING: rtpproxy [rtpproxy.c:1432]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily > 14(1596) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (9): debugger > 5(1584) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (5): nathelper > 14(1596) DEBUG: debugger [debugger_mod.c:124]: rank is (9) > 5(1584) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (5): rtpproxy > 5(1584) ERROR: rtpproxy [rtpproxy.c:1466]: can't connect to RTP proxy > 5(1584) ERROR: rtpproxy [rtpproxy.c:1538]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it > 5(1584) WARNING: rtpproxy [rtpproxy.c:1395]: can't get version of the RTP proxy > 5(1584) WARNING: rtpproxy [rtpproxy.c:1432]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily > 5(1584) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (5): debugger > 5(1584) DEBUG: debugger [debugger_mod.c:124]: rank is (5) > 15(1597) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (10): rtpproxy > 15(1597) ERROR: rtpproxy [rtpproxy.c:1466]: can't connect to RTP proxy > 15(1597) ERROR: rtpproxy [rtpproxy.c:1538]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it > 15(1597) WARNING: rtpproxy [rtpproxy.c:1395]: can't get version of the RTP proxy > 15(1597) WARNING: rtpproxy [rtpproxy.c:1432]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily > 15(1597) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (10): debugger > > > > paste the rtpproxy/nathelper module params lines which are in your > kamailio.cfg. > > regards, > sammy > > > On Wed, Jan 4, 2012 at 7:59 PM, nunu abe <nunu_abe(a)yahoo.com> wrote: > > Thank you Sammy for noticing :) > > Now I have a bit of different but same problem. After the earlier posting > the message I decided to start everything from scratch and I installed > everything new. > But I still have problem. The rtpproxy server is not working, it fails to > replace the IP address of the sender in the SDP and hence there is no RTP > communication. > These are the steps I followed: > > > 1. Freshly installed kamailio 3.2.1 on Kubuntu from GIT > > 2. Installed mysql server > > 4. installed rtpproxy, changed its user to be kamailio. > > 3.Copied the provided kamailio.cfg file from the link and edited the IP > addresses and the DBURL to reflect my settings > > 4. Uncommented the Control socket line and added these lines in the > /etc/default/rtpproxy file > CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" > ADDR_IPV4=[10.10.10.10] > ADDR_IPV6=[3001:0:0:1::10] > EXTRA_OPTS="-l {ADDR_IPV4} -6 /{ADDR_IPV6}" > > *** I aslso tried without adding the last three lines, but it made no > difference.*** > > 5. Started rtpproxy in a bridge mode > > xxxx@DualStackCS:~$ sudo rtpproxy -F -l 10.10.10.10 -6 /3001:0:0:1::10 > *** I have also tried to start it this way :xxxx@DualStackCS:~$ sudo > rtpproxy -F -l 10.10.10.10 -6 /3001:0:0:1::10 -s > unix:/var/run/rtpproxy/rtpproxy.sock - but no difference**** > > xxxx@DualStackCS:~$ ps aux|grep rtpproxy > root 1522 0.0 0.0 11384 696 ? Ssl 15:31 0:00 rtpproxy > -F -l 10.10.10.10 -6 3001:0:0:1::10 > xxxx 1528 0.0 0.0 3324 804 pts/1 S+ 15:31 0:00 grep > --color=auto rtpproxy > > xxxxx@DualStackCS:~$ sudo /etc/init.d/kamailio start > > > 6. Started kamailio > @DualStackCS:~$ sudo /etc/init.d/kamailio restart > > I have attached the log file. > > Many thanks for your help. > > Regards, > > Maedot > > *From:* Sammy Govind <govoiper(a)gmail.com> > *To:* miconda(a)gmail.com; SIP Router - Kamailio (OpenSER) and SIP Express > Router (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> > *Sent:* Tuesday, January 3, 2012 2:04 PM > *Subject:* Re: [SR-Users] RTPproxy on Kamailio 3.2.1 difficulty. > > Hi, > > I had a good laugh when I read the reply, but going through the whole > thread again I found this: > " It displays the same message even when I replace ADDR_IPV4 and > ADDR_IPV6 with IP addresses. " > > So there _IS_ some problem here. > > Nuno, can you please copy paste the exact command which you executing, > plus paste the logs, > > -- > Regards, > Sammy. > > On Tue, Jan 3, 2012 at 2:22 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > > Hello, > > the ADDR_IPV4 and ADDR_IPV6 have to be replaced with real IP addresses > that you have on the system runnin RTPProxy. > > Cheers, > Daniel > > > On 1/3/12 8:56 AM, nunu abe wrote: > > Dear Daniel, > Greetings and Happy New year to you all! I would like to thank you very > much for your detailed and well structured tutorials on Kamailio. I am > resending this email as I sent the earlier one before I was a member of the > mailing list and I thought it may have been sent to the spam folder. If not > then, I apologize for double posting. > I was trying to design a similar scenario like the one " Run your own > SIP VoIP service on both IPv4 and IPv6 " as explained on Asipto web-page( > http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6). However I ran > into difficulties with rtpproxy. The proxy is not routing media packets > as it should. I followed the instruction regarding the kamailio.cfg file > and edited the necessary lines according to my settings. I think my problem > comes from setting up the rtpproxy. > 1. I edited the /etc/default/rtpproxy file. i.e. I uncommented the line > - CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" > 2. I started the rtpproxy with this command: rtpproxy -F -l ADDR_IPV4 -6 > /ADDR_IPV6 . However, after this command the system returns the message : rtpproxy: > setbindhost: No address associated with hostname > It displays the same message even when I replace ADDR_IPV4 and ADDR_IPV6 > with IP addresses. > > 3. I have also tried using the udp socket:- CONTROL_SOCK=udp: > 127.0.0.1:22222, of course I have replaced the unix socket by this udp > socket in the kamilio.cfg file. > So my question is, what did I do wrong with the settings that SIP > messages are traversing normal from IPv4 client to IPv6 client and vice > versa but media files(RTP packets) are not? The rtpproxy receives the media > files from one client but fails to forward them to the other client. > Thanks for your help. > Maedot > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda > > > _______________________________________________ > 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 > > > > _______________________________________________ > 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 > > > > _______________________________________________ > 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 > > > > > > _______________________________________________ > 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 > > > > > >

2 8

most links broken on http://www.iptel.org/ser/doc/gettingstarted host siprouter.teigre.com is unknown)
by Klausfpga 10 Jan '12

10 Jan '12

Hi, II just want to find the minimalistic setup to get a sip registrar up and running. I went to http://www.iptel.org/ser/doc/gettingstarted and tried to download the html documentation: ( http://siprouter.teigre.com/doc/gettingstarted/ ) My browser reports that this host is unknown all the other links are also pointing to the same unknown host. Has the server been renamed. Having the 'Getting Started' Software not available renders ser useless for all new users. I used ser already many years ago, with a very simple config file and if I remember well, there were some very primitive example config files for simple use cases, but somehow I don't know how to locate them. I'd like to have a minimalistic setup for a sip registrar with softphones connected on two subnets (no NAT or firewall involved) and am rather stuck without some documentation. Thanks in advance for any pointers

3 2

app_lua coredump
by noc＠nelcom-voip.com 10 Jan '12

10 Jan '12

I've installed kamailio (3.2, 3.2-nightly, dev-nightly) for debian unstable (wheezy) using repository at http://deb.kamailio.org and constantly get a coredump with this lua code: -- SIP request routing function route_request(name) sr.dbg("routing SIP request from Lua [" .. name .. "]\n") -- initial checks of SIP request if sr.maxfwd.process_maxfwd(10) < 0 then sr.sl.send_reply(483,"Too Many Hops") return end end I've started with an example at this page http://kb.asipto.com/kamailio:usage:k32-lua-routing, and then just reduced the lua script to code above. If I do comment sr.maxfwd.process_maxfwd(10) it passes. It is impossible to call sr.modf("mf_maxfwd...") because it says about fixups. What's wrong? -- With best regards, Roman Galeev.

2 14

Prepaid billing solution for a service provider
by Carlo Dimaggio 09 Jan '12

09 Jan '12

Hi all, I'm working in a project for a service provider in which prepaid is an essential requirement. They have about 12000 subscribers. The core infrastructure will be Kamailio+RTPProxy while I have some doubts about the prepaid feature. I'm thinking about a B2BUA (SEMS or Freeswitch) that is called if the user belong to a "prepaid group" and perform authorization and accounting; in this case the SIP flow could be: Kamailio -> B2BUA(Prepaid) -> Kamailio. Is a good choice? At the moment I cannot estimate the effort needed to develop this section (base prepaid feature) of the project... Do you have some hints? What are the prepaid solution implemented by you? Do you use your own developed solution or do you think that is better to choose a commercial solution (that provide also calling-card and other features)? I have seen some commercial products (Portabilling, Jerasoft/Bilberry, ...); what do you think about? Best Regards,

6 8

Kamctl reload commands
by Uri Shacked 09 Jan '12

09 Jan '12

Hi, i am using different modules like "mtree", "carrierroute", "drouting" and so. how can i tell how long did the reload takes? how do i know if it fails? is there a way to get a return code from kamailio that confirms that the reload ended and successfuly? BR, Uri

3 4

Re: [SR-Users] Topoh Log Warning
by Daniel-Constantin Mierla 09 Jan '12

09 Jan '12

Hello, On 1/6/12 10:50 PM, Javier Vidal wrote: > Hi Daniel, problem solved!!! thanks, was a bad REGISTER, i was wrong. > > Thank you!!!! welcome, jut keep the mailing list always on cc, to help others to look for solutions of similar cases. Cheers, Daniel > > El 06/01/2012, a las 19:34, Daniel-Constantin Mierla escribió: > >> Hello, >> >> On 1/6/12 6:47 PM, Javier Vidal wrote: >>> Hi all, >>> >>> Two days ago, i had activated topoh module, and now in log file, when i receive a Options, appears this error >>> >>> /usr/sbin/kamailio[29936]: WARNING: topoh [th_msg.c:1063]: Invalid/Unparsed CSeq in message. Skipping. >>> >>> I dont' know solve it, only deactivating the topoh module!! >> does the OPTIONS have the CSeq header? Can you paste here an ngrep trace of such OPTIONS? >> >> The warning message is printed from a function that tries to detect whether is a REGISTER or PUBLISH, which are requests that don't get masked since they end to a service. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.com >> http://linkedin.com/in/miconda -- http://twitter.com/miconda >> -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Topoh Log Warning
by Javier Vidal 06 Jan '12

06 Jan '12

Hi all, Two days ago, i had activated topoh module, and now in log file, when i receive a Options, appears this error /usr/sbin/kamailio[29936]: WARNING: topoh [th_msg.c:1063]: Invalid/Unparsed CSeq in message. Skipping. I dont' know solve it, only deactivating the topoh module!! Sorry for my english. ;) Regards. Javier Vidal

2 1

RTPproxy on Kamailio 3.2.1 difficulty.
by nunu abe 06 Jan '12

06 Jan '12

Dear Daniel, Greetings and Happy New year to you all! I would like to thank you very much for your detailed and well structured tutorials on Kamailio. I am resending this email as I sent the earlier one before I was a member of the mailing list and I thought it may have been sent to the spam folder. If not then, I apologize for double posting. I was trying to design a similar scenario like the one " Run your own SIP VoIP service on both IPv4 and IPv6 " as explained on Asipto web-page(http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6). However I ran into difficulties with rtpproxy. The proxy is not routing media packets as it should. I followed the instruction regarding the kamailio.cfg file and edited the necessary lines according to my settings. I think my problem comes from setting up the rtpproxy. 1. I edited the /etc/default/rtpproxy file. i.e. I uncommented the line - CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" 2. I started the rtpproxy with this command: rtpproxy -F -l ADDR_IPV4 -6 /ADDR_IPV6 . However, after this command the system returns the message : rtpproxy: setbindhost: No address associated with hostname It displays the same message even when I replace ADDR_IPV4 and ADDR_IPV6 with IP addresses. 3. I have also tried using the udp socket:- CONTROL_SOCK=udp:127.0.0.1:22222, of course I have replaced the unix socket by this udp socket in the kamilio.cfg file. So my question is, what did I do wrong with the settings that SIP messages are traversing normal from IPv4 client to IPv6 client and vice versa but media files(RTP packets) are not? The rtpproxy receives the media files from one client but fails to forward them to the other client. Thanks for your help. Maedot

3 8

Kamailio 3.2 RTP repacketization
by Arif Tuhin 06 Jan '12

06 Jan '12

Hi,I'm using the new kamailio 3.2 which has some changes in rtpproxy module. And i'm finding it hard to implement a certain configuration. Because of our network infrastructure i have to repackage rtp streams in one direction. Now in kamailio 3.1.5 i used to do this in this way:In main routeroute[RTPPROXY]{if(is_method("BYE"){ unforce_rtp_proxy();else if(is_method("INVITE")){ force_rtp_proxy();}in onreply_route force_rtp_proxy("z160"); Now in kamailio 3.2 the force_rtp_proxy() is replaced by rtpproxy_manage(). Here the script reads like:main routeroute[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("z150"); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; onreply_routethis block is called route[NATMANAGE]. Now if i put rtpproxy_manage("z150"), it tries to repackage in both direction which i do not want. So i need to write a separate block for onreply_route instead of calling route[NATMANAGE]. So please help me to write a block for onreply_route.Thanks in advance. With Best Regards Ariful Hossain Tuhin email: 1. etothepowerpi(a)gmail.com 2. etothepowerpi(a)hotmail.com 3.etothepowerpi(a)yahoo.com skype: freeburn1986

2 1

auth module realm_prefix
by Kelvin Chua 06 Jan '12

06 Jan '12

Hi guys, just wondering, what happened to realm_prefix on auth module for 3.2.x? why was it removed?

2 3

dialog module - set_dlg_profile won't work
by Ben WIlliams 05 Jan '12

05 Jan '12

Hi, can anyone see why this code wouldn't work? In the log is shows this message after set_dlg_profile called - dialog [dlg_hash.c:523]: no dialog callid='...' found Dialogs are being saved to dialog table ok but nothing is in dialog_vars table. modparam("dialog", "dlg_flag", DLGFLAG) modparam("dialog", "hash_size", 128) modparam("dialog", "default_timeout", 14400) modparam("dialog", "dlg_match_mode", 0) modparam("dialog", "table_name", "dialog") modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 2) modparam("dialog", "db_update_period", 10) modparam("dialog", "profiles_with_value", "caller ; allcalls") ... if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting setflag(DLGFLAG); ## add to dialog list $var(callcnt) = 0; get_profile_size("caller", "$fu", "$var(callcnt)"); xlog("$fu has $var(callcnt) concurrent calls\n"); if ($var(callcnt) >= 2) { sl_send_reply("503", "Simultaneous calls limit reached"); exit; } set_dlg_profile("caller","$fu"); } -- Thanks Ben

2 2

No memory left in send_subscribe (PUA) from update_pua while building the tm dlg_t structure
by laura testi 04 Jan '12

04 Jan '12

Hi, we are using the PUA_XMPP and PUA modules from the master branch. When the modules are started, everything are ok, the presence events from XMPP are sent to kamailio SIP servers (PUBLISH/SUBSCRIBE) and cached in the hash. But when there are several thousands records in the hash tabel, the update_pua function called in the hashT_clean gives a lot of "No memory left" error when the hashT_clean is waked up from the time: ERROR: pua [send_subscribe.c]: No memory left ERROR: pua [pua.c]: while building tm dlg_t structure The failed call is: td = (dlg_t*)pkg_malloc(size); if(td == NULL) { LM_ERR("No memory left\n"); return NULL; } in dlg_t* pua_build_dlg_t(ua_pres_t* presentity) function in send_subscribe.c. The size is about 400 and something... It's strange..... Is it the memory leak in the PUA module? I also try to increase the pkg_memory from 4MB default to 16MB, but it doesn't help. Any Idea? Thanks in advanced Laura

2 14

Authentication Feature Question
by Ali Jawad 04 Jan '12

04 Jan '12

Hi I do have 3 Kamailio servers, one for mobile phone registrations, one for softphone registrations and one for SIP device registrations. Each of those devices connects to it's perspective kamailio server sip1.domain.com sip2.domain.com sip3.domain.com All 3 Kamailio servers share the same database, and users can use their kamailio user/pwd on any of the devices, now I want to use encrypted passwords and remove clear text passwords from the database. I did test with one server and all is fine,however if a user want to register from the second kamailio server it does not work, basically because the db domain entry from which the hash is created is sip1.domain.com and stored in the db, while the user connects from to sip2.domain.com this eventually generates a different hash. Is there anyway to overcome this ? Can I exclude Domain from Hash generation ? Any other option that allows me to do the above ? Thanks

2 20

Intermittent start up issue
by Ali Jawad 03 Jan '12

03 Jan '12

Hi All I have tried the below with Kamailio 3.1.0 and 3.2.1. the problem is that Kamailio at times fails to start up for no obvious reasons, I.e. in 10 restarts Kamailio might fail to start 3,0 or 5 times. Te debug log "level 2 " shows the following : 27(24762) ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on 'xxxxxxxxx' (4) 27(24762) ERROR: <core> [db.c:303]: could not add connection to the pool27(24762) ERROR: group [group.c:211]: unable to connect to the database 27(24762) ERROR: <core> [sr_module.c:889]: init_mod_child(): Error while initializing module group (/usr/local/kamailio/lib/kamailio/modules_k/group.so) 27(24762) ERROR: <core> [pt.c:481]: ERROR: fork_tcp_process(): init_child failed for process 27, pid 24762, "tcp receiver child=1" and then 30(24765) : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 45 0(24735) ALERT: <core> [main.c:748]: child process 24762 exited normally, status=255 0(24735) INFO: <core> [main.c:766]: INFO: terminating due to SIGCHLD The problem happens on 4 different servers, I did try to compile more than once, the servers are 32 Bit CentOS 5.6. The connection to the database is monitored and there are no issues with connecting to the DB "it works at times and times not for Kamailio so a user/pwd issue is not likely to cause this, manual cnnect worked", config file errors are not found. I did also check max no of connections on MySQL server which is set to 500 on the server and only 100 cons are used. The group module uses the same connection string as other modules which is predefined and there are other modules that access the db before the group module and those do not have a problem. Please advice. Regards

4 15

New Year, New Module: app_mono - embedded execution of managed code
by Daniel-Constantin Mierla 03 Jan '12

03 Jan '12

Hello, the first committed module in 2012 is app_mono, which offers embedded execution of manage code (e.g., C#/.NET) via Mono project (http://www.mono-project.com/). The readme of the new module is available at: * http://kamailio.org/docs/modules/devel/modules/app_mono.html Current API which exported by SIP server for usage in C# application is documented at: * http://www.kamailio.org/wiki/embeddedapi/devel/mono Besides C#, app_mono should be able to run managed code compiled from applications written in other languages such as VisualBasic.NET, Java, Java Script, Python, ... more are listed at: * http://www.mono-project.com/Languages A primary use is integration with several widely used Microsoft technologies and APIs, for example C# having up-to-date libraries to connect to Active Directory LDAP service or MS SQL Server. Happy 2012, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

usrloc, timer process and cache cleanup
by Andreas Granig 02 Jan '12

02 Jan '12

Hi, Could you please tell me which of the three timer processes ("timer", "slow timer" or "timer nh") is responsible for cleaning up the internal usrloc cache? Looks like every now and then the cleanup of the internal location cache is starting to fail. Funny thing is that expired locations are removed from the mysql backend, but not from the internal cache. We're running kamailio 3.1.5, are there any known issues fixed since that version? In the meanwhile we're trying to pin the issue down, but maybe someone has a clue... Andreas

5 11

Problems with video behind NAT
by Stoyan Mihaylov 02 Jan '12

02 Jan '12

My whole configuration is: [Sip clients] < = > Kamailio 3.2 <=> Asterisk servers (behind Kamailio) Asterisk servers have only local IP addresses, and I use t_relay instead of forward. Kamailio runs on same server as rtpproxy. Now I played lot of combinations and I have next results: 1. Clients are in same net with Kamailio server (and Asterisk server). Client call Kamailio, Kamailio t_relay to Asterisk, Asterisk dial back to Kamailio, and Kamailio dial Client. rtpproxy is working. Video and audio are fine... 2. Same as above, but clients are behind NAT. Sound is perfect, no video. 3. Same as in point 1, but no rtpproxy, and no Asterisk servers. Sound and video are perfect. With rtpproxy - no voice, no video. I would like to find solution for point 2. Sound and video are on different rports (seen in wireshark). As I understand, my error is either in wrong forwarding of ACK (I do t_relay to Asterisk in case from and to addresses are same), or may be I have to find correct FLAGS for rtpproxy_manage(), or even replace it and place with rtpproxy_offer (answer etc). Bellow is my kamailio.cfg: ####### Defined Values ######### #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_IPAUTH #!define WITH_NAT #!define WITH_PSTN # *** Value defines - IDs used later in config #!define DBURL "mysql://user:pass@192.168.2.251/openser" #!define MULTIDOMAIN 0 #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### debug=3 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 port=5060 tcp_connection_lifetime=3605 #!ifdef WITH_PSTN pstn.gw_ip = "192.168.2.251" desc "PSTN GW Address" #!endif # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="//lib64/kamailio/modules_k/://lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif loadmodule "htable.so" #!ifdef WITH_ANTIFLOOD loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- loadmodule "dispatcher.so" modparam("dispatcher", "db_url", DBURL) # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "//etc/kamailio/tls.cfg") #!endif modparam("htable", "htable", "forw=>size=8;autoexpire=7200;") #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif request_route { xlog("L_ALERT","Pakage $rm from $fu (IP:$si:$sp)\n"); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); route(ACKBYE); # handle requests within SIP dialogs route(WITHINDLG); if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } route(SIPOUT); route(PRESENCE); route(REGISTRAR); if ($rU==$null) { sl_send_reply("484","Address Incomplete"); exit; } route(PSTN); route(LOCATION); route(RELAY); } route[RELAY] { if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } route[ACKBYE] { #!ifdef WITH_PSTN if (is_method("BYE|ACK")) { xlog("L_ALERT","AB $rm $sht(forw=>$ft) $td"); if(($sht(forw=>$ft))=~"MessageCPIM"){ # Direct messages between clients xlog("L_ALERT","AB $rm CPIM $td"); return; } if(src_ip==$td){ #I have to rewrite du - messages loop in Kamailio xlog("L_ALERT","ACK,Bye Method equalIP"); $du=$sht(forw=>$ft); xlog("L_ALERT","ACK,Bye $ft $du"); route(RELAY); exit; } xlog("L_ALERT","ACK,Bye Not me"); } #!endif return; } route[PSTNINVITE] { if(is_method("INVITE")){ if($rb=~"message/CPIM"){ # Direct messages between clients $sht(forw=>$ft)="MessageCPIM"; route(LOCATION); route(RELAY); } #This way I can select from multiple Asterisk servers. I will t_relay instead of forward, because Asterisk servers are with local IP. ds_select_dst("1","4"); $sht(forw=>$ft)=$du; sl_send_reply("100","Trying"); route(RELAY); exit(); } } route[PSTN] { #I am using this routine to forward to Asterisks (instead of additional one for balancing) #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } if(allow_source_address()) return; if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if(is_method("MESSAGE")) return; route(PSTNINVITE); route(RELAY); exit; #!endif return; } # route to voicemail server route[TOVOICEMAIL] { return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif }

2 1

Happy New Year!
by Daniel-Constantin Mierla 31 Dec '11

31 Dec '11

This was it for 2011, final timeout for a fruitful year! Many thanks to the community for keeping things moving forward! A happy and prosperous 2012 to all Kamailio and SER friends!!! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

problem with create and release tcp connections
by Bruno Bresciani 29 Dec '11

29 Dec '11

Hi all, Someone know if the kamailio 1.5.0 has problem to create and release tcp connetctions after some time running? In my SIP network I have six IP telephones and one gateway registered over TLS, after some time (one day) this devices have problems to register and make calls. I noticed through the log a freezing of kamailio when it try release the tcp connections. Below is part of the log where I notice this behavior: *Dec 28 09:46:46 vts74 /home2/local/kamailio/sbin/kamailio[8951]: DBG:core:io_watch_add: io_watch_add(0x816cc80, 25, 2, 0xb61e6558), fd_no=20 Dec 28 09:46:46 vts74 /home2/local/kamailio/sbin/kamailio[8951]: DBG:core:handle_tcp_child: cmd CONN_RELEASE 0xb61e6558 refcnt= 0 Dec 28 09:47:11 vts74 /home2/local/kamailio/sbin/kamailio[8940]: DBG:usrloc:nodb_timer: Binding '2503','sips:2503@192.168.174.178;transport=tls' has expired Dec 28 09:47:26 vts74 /home2/local/kamailio/sbin/kamailio[8951]: DBG:core:handle_tcpconn_ev: data available on 0xb621d710 30 Dec 28 09:47:26 vts74 /home2/local/kamailio/sbin/kamailio[8951]: DBG:core:io_watch_del: io_watch_del (0x816cc80, 30, -1, 0x0) fd_no=21 called * I know that upgrade to kamailio 3xx is recommended and I'm working to do it, but before, I have to understand this behavior. Cheers

2 2

Pseudo-variables for "Status-Line"
by Robert R 29 Dec '11

29 Dec '11

Hi, What is the Pseudo-variables for "Status-Line" filed of SIP response messages (2xx, 3xx,4xx,5xx,6xx)? i.e., is there a Pseudo-variables to display the SIP response code? Thanks, R

4 5

Dispatcher does not trigger the event_route
by Mino Haluz 29 Dec '11

29 Dec '11

Hi, I have two gateways pinged by kamailio. The both are AP (active/probing), when I cut the one gateway off, it becomes IP(inactive/probing) but the event_route is not fired up. Am I missing something ? event_route[dispatcher:dst-down] { xlog("L_ERR", "Destination down: $rm $ru ($du)\n"); } Mino

3 5

Load Balancing
by Ronald Kiel 28 Dec '11

28 Dec '11

Hi! Relatively new here, so forgive my probably newbie question.. I understand Kamailio has the Dispatcher-module for load-balancing. OpenSIPS has both a Dispatcher-module and a LB-module, which seems to offer improved functionality. Is there also something like the LB-module in Kamailio? Cheers, Ron

3 2

ACK From an OK it's being discarded
by Lucas Alvarez 27 Dec '11

27 Dec '11

I have Kamailio 3.2.0 between two asterisk servers, after the call set, one of the kamailio send the OK from the INVITE and the return ACK of that message was discarded. This makes asterisk hangup the call after 5 secs. It's that right? OK message: U 172.25.249.15:5060 -> 172.25.249.14:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.25.249.14:5060;branch=z9hG4bK09fc3de6;rport=5060. From: "asterisk" <sip:asterisk@172.25.249.14>;tag=as6411602a. To: <sip:775008@172.25.249.15:5060>;tag=as55ab3180. Call-ID: 547225391b7828402ecaa03e1dab5a86(a)172.25.249.14. CSeq: 102 INVITE. Server: Asterisk PBX 1.8.7.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: <sip:775008@172.25.249.15:5080>. Remote-Party-ID: "Eus Test" <sip:3999@172.25.249.14 >;party=called;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 285. . v=0. o=root 2045590031 2045590031 IN IP4 172.25.249.15. s=Asterisk PBX 1.8.7.1. c=IN IP4 172.25.249.15. t=0 0. m=audio 11922 RTP/AVP 0 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. Discarded ACK: U 172.25.249.14:5060 -> 172.25.249.15:5060 ACK sip:775008@172.25.249.15:5080 SIP/2.0. Via: SIP/2.0/UDP 172.25.249.14:5060;branch=z9hG4bK6ea5aff6;rport. From: "asterisk" <sip:asterisk@172.25.249.14>;tag=as6411602a. To: <sip:775008@172.25.249.15:5060>;tag=as55ab3180. Contact: <sip:asterisk@172.25.249.14>. Call-ID: 547225391b7828402ecaa03e1dab5a86(a)172.25.249.14. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Remote-Party-ID: "asterisk" <sip:asterisk@172.25.249.14>. Content-Length: 0. . Kamailio's configuration where the ACK message it's being discarded: if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } It would be ok if I relay the ack even if it didn't match any transaction?? Any help would be appreciated. Regards, Lucas

2 1

lcr module and IPv6
by Andreas Granig 26 Dec '11

26 Dec '11

All, Just realized that lcr module does not support IPv6 yet. Is there already any work going on here? Otherwise we'd take this task over. Andreas

3 9

Merry Christmas!
by Daniel-Constantin Mierla 24 Dec '11

24 Dec '11

I want to take the opportunity and give proper credits to everyone around the project, any bit of contribution makes it evolve better and better year by year. This includes also the involvement in community, we don't need only developers to improve. There is no reason to stay aside in an open source project, become active and play a role in its evolution. We've just started the second decade of development, but there is always space to innovate communications. With SER and Kamailio, we pioneered over the years many solutions in unified communication systems, the plans are the same, be ahead of the market with new features while preserving the robustness. Merry Christmas and great winter holidays to all Kamailians and SERians!!! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 1

How to get Reason code from reply in failure_route
by Mino Haluz 23 Dec '11

23 Dec '11

Hi, how can I get the reason code from reply in failure_route ? $hdr(reason) points to the INVITE request.. Thanks, Mino

2 3

kagoor voiceflow 4000
by 3samnet 22 Dec '11

22 Dec '11

Hi, I am Isam; I have bought used Juniper vf4000, without any (manuals, datasheets, disc, etc.). So maybe you something like that or you has an idea how to configure it. At least to reset the system, because I don't have login or password. Kind Regards, Isam Awadallah

1 0

Music on hold with freeswitch as SBC
by Gautam Batra 22 Dec '11

22 Dec '11

Hello, I have a kamailio sip proxy server with freeswitch acting as SBC. I want to redirect the call to freeswitch when hold is pressed so that i can play music on hold. I tried this by using rewritehostport in case of a re-invite, but the call drops in that case. Could someone please help me with this? Gautam

3 9

ERROR: slow timer too slow: overflow (1048 - 25 = 1023) repeat a lot
by laura testi 22 Dec '11

22 Dec '11

Hi, sometime we get the following error after we restart the kamailio for a while: ERROR: <core> [timer.c:972]: ERROR: slow timer too slow: overflow (1048 - 25 = 1023) ... last message repeated 138times .... ERROR: <core> [timer.c:972]: ERROR: slow timer too slow: overflow (1048 - 25 = 1023) last message repeated 49 times it print them continuously.... is it normal? How I can fix it? Thank you very much! Best Regards, Laura

2 2

anyone with opensuse around?
by Daniel-Constantin Mierla 22 Dec '11

22 Dec '11

Hello, is anyone here having an opensuse available for using to do some tests? It is about the radius modules, it seems opensuse does now have radiusclient-ng, but the client library from freeradius. They should be quite similar and we should be able to compile with freeradius client library, like make FREERADIUS=1 ... I tried on opensuse build service and it fails due to some conflict on DBG definition. It is hard to troubleshoot on OBS, so maybe someone can give a hand and find the fix faster -- I can provide guidelines or if I can get shell access (just simple user) would be better (write me directly in this case). The goal is to get the radius modules packaged as rpm on opensuse. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

new in devel wiki page
by Daniel-Constantin Mierla 22 Dec '11

22 Dec '11

Hello, short reminder for everyone to update the wiki page with short description of the new stuff/enhancements added to GIT repository -- here it is: * http://www.kamailio.org/wiki/features/new-in-devel Also, if you do changes to existing parts that require upgrade of config file or database schema from v3.2.x to devel, please write short guidelines at: * http://www.kamailio.org/wiki/install/upgrade/3.2.x-to-3.3.0 Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] Who can tell me how to get the dialog-based CDRs ?
by Timo Reimann 21 Dec '11

21 Dec '11

Hey there, I added the list to CC as others may be interested in your findings as well. "sunny_day" <416415455(a)qq.com> schrieb: > Thank you very much for your detailed analysis ! > According to your analysis I have been getting more understand the > kamailio. :-) > After submitting the issue, I have done many experiments. And now, I > may find the reason about my script. > > 1. somebody seems like having said dialog is tracing and dealed with > after loose_route()--(May be I didn't understand it correctly. And now > I am also not sure the relationship between dialog and loose_route() ). > And the dialog will be destroyed after receiving the "BYE" message. > So, I place the "insert" after loose_route() and "is_method("BYE")" . > ------it may have problem. Calling loose_route() makes sure that for a given sequential request (which includes BYE messages) it is properly linked to its dialog. This is carried out by either adding cookies to Route headers or trying to match the dialog-identifying SIP headers (From-Tag, To-Tag, and Call-ID). This so-called "matching mode" is set to "DID only" which corresponds to using cookies in Route headers. For this to work, you must ensure that calls are being enriched with Record-Route headers. So one possible reason why things didn't work for you is that you used "DID only" matching mode (which you apparently did as you configuration didn't deviate from the default) but forgot to add Record-Route headers. Alternatively, you could set matching mode to 2 (SIP header-based matching only) or 1 (cookie-based but fallback to header-based) to get things going dialog-wise. See the dlg_match_mode modparam documentation here: http://sip-router.org/docbook/sip-router/branch/master/modules_k/dialog/dia… . > The main reason "insert" can not insert the data into cdr table is > loose_route() returns "0", which has no "Route:" header , and the > "insert" will never be executed. Which may be the case because you did not instruct Kamailio to add Record-Route headers along the signaling way. > 2. At last, I place the "insert" in the "route(RELAY);". And data can > be inserted into the cdr table again. Originally, I was considering the > dialog will be destroyed immediately when then "BYE" message received. > The results of my experiments prove that it isn't. But I am not sure > whether my conclusion is right. In the old days, dialogs would indeed be destroyed as soon as the transaction correlating to the first BYE request was deleted. This was changed a while in ago in order to be able to handle responses to such BYE requests too. So these days, dialogs do not get destroyed until a BYE-completing response is seen. (More specifically, until the transaction related to a BYE-completing response is terminated. Or, if that BYE request never sees a response and its transaction times out, whichever comes first. But that's just details.) You probably shouldn't be satisfied with updating database entries during route[RELAY] (at least not only). If it's not working on processing of a final BYE request, this indicates that dialog tracking isn't properly working for you (see my explanation above as to why). Whatever you store in dialog variables during an INVITE may not be valid by the time you persist to database. And regarding CDR generation using Kamailio's built-in mechanism: It's part of the acc (accounting) module (not dialog as I initially claimed). You may want to look at the documentation to see if it's a fit for you: http://sip-router.org/docbook/sip-router/branch/master/modules_k/acc/acc.ht… HTH, --Timo > ------------------ Original ------------------ > From: "Timo Reimann"<sr(a)foo-lounge.de>; > Date: Wed, Dec 21, 2011 06:51 PM > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List"<sr-users(a)lists.sip-router.org>; > Cc: "sunny_day"<416415455(a)qq.com>; > Subject: Re: [SR-Users] Who can tell me how to get the dialog-based > CDRs ? > > > Hey, > > > Am 21.12.2011 um 07:03 schrieb sunny_day: >> My object: get the dialog-based CDRs >> >> SIP proxy: kamailio-3.2.0 >> >> My script: >> >> loadmodule "dialog.so" >> ...... >> modparam("dialog","dlg_flag",4)#Must be set to create the dialog > associated to an initial request. >> modparam("dialog","db_url","mysql://xxx:xxx@localhost/albert") >> modparam("dialog","table_name","dialog") >> ....... >> request_route{ >> if( is_method("INVITE") && !has_totag()) >> { >> dlg_setflag(4); >> dlg_var(start_time)=$TS; >> dlg_var(caller)=$fU; >> dlg_var(caller)=$tU; >> } >> dlg_manage(); >> ...... >> # dispatch destinations to PSTN >> route(PSTN); >> # user location service >> route(LOCATION); >> route(RELAY); >> } >> ......... >> route[WITHINDLG] { >> if (has_totag()) { >> # sequential request withing a dialog should >> # take the path determined by record-routing >> if (loose_route()) { >> if (is_method("BYE")) { >> setflag(FLT_ACC); # do accounting ... >> setflag(FLT_ACCFAILED); # ... even if the > transaction fails >> >> sql_query("cd", "insert into > cdr(caller,callee,start_time,duration) > values($dlg_var(caller),$dlg_var(callee),$dlg_var(start_time),$DLG_lifetime)", > "rd"); >> sql_result_free("rd"); >> } >> if (is_method("ACK")) { >> route(NATMANAGE); # ACK is forwarded statelessy >> } >> route(RELAY); >> } >> ........ >> >> >> My issue is : >> >> The dialog records can be wrritten into the dialog table. > But , the sql_query can not executed. So the cdr table will not get the > CDRs. I don't know why. Who can help me ? >> This is a simple script for testing. >> >> Any suggestion will be appreciated. >> >> Thank you in advance. > > It's very hard to help with any Kamailio error unless you provide the > actual error message. Whatever the issue is with the sqlops module, > Kamailio will tell you, so please tell us. > > Until then, I can only speculate what's gone wrong: Did you possibly > forget to specify the sqlcon module parameter? You need to setup the > MySQL connection parameters in there. See > > http://sip-router.org/docbook/sip-router/branch/master/modules_k/sqlops/sql… > > for sqlcon's documentation and > > http://sip-router.org/docbook/sip-router/branch/master/modules_k/sqlops/sql… > > for the reference regarding sql_query(). Please note how that part > states that "the result parameter should normally only be omitted when > no result is expected (INSERT, UPDATE, DELETE)" (which is the case in > your test script). In consequence, you won't need to free the result > set which could be another source of error. > > > Finally, please note that the dialog module itself is capable of > producing CDRs directly from the dialog machinery. It's yet lacking > support for direct output to a database (AFAIK), however, so you'd have > to post-process the CDR results (which are being written to file as of > now) in order to get them persisted into a database. > For some reason, the CDR documentation I wrote seems to have > disappeared from the git repository. I'll investigate that further and > provide you with a link to the dialog-way-of-creating-CDRs when > available (and desired from your side). > > > HTH, > > --Timo

1 0

carrierroute issue with rewrite host
by Uri Shacked 21 Dec '11

21 Dec '11

Hi, I am using kamailio (only one) as routing engine. I have 2 gateways. Both of them sending calls to kamailio. When i use the carrierroute module i need to change (and succeed) the user ($rU). But, i need to send the call (the invite) back to the gateway that sent me the call from the beginning (it can be either one of those 2). So, my problem is that i don’t want to rewrite the host in carrierroute, just the user. If i leave the rewrite_host field empty, it is considered to be "black list" and drops the call. If i rewrite the host, i can send the call to the second gateway that was not the one that sent the call, and the call drops as well. any ideas how to use carrierroute for changing user only and send the invite back to the gateway that originally sent the call? Thanks, Uri

3 2

Who can tell me how to get the dialog-based CDRs ?
by sunny_day 21 Dec '11

21 Dec '11

My object: get the dialog-based CDRs SIP proxy: kamailio-3.2.0 My script: loadmodule "dialog.so" ...... modparam("dialog","dlg_flag",4)#Must be set to create the dialog associated to an initial request. modparam("dialog","db_url","mysql://xxx:xxx@localhost/albert") modparam("dialog","table_name","dialog") ....... request_route{ if( is_method("INVITE") && !has_totag()) { dlg_setflag(4); dlg_var(start_time)=$TS; dlg_var(caller)=$fU; dlg_var(caller)=$tU; } dlg_manage(); ...... # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } ......... route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails sql_query("cd", "insert into cdr(caller,callee,start_time,duration) values($dlg_var(caller),$dlg_var(callee),$dlg_var(start_time),$DLG_lifetime)", "rd"); sql_result_free("rd"); } if (is_method("ACK")) { route(NATMANAGE); # ACK is forwarded statelessy } route(RELAY); } ........ My issue is : The dialog records can be wrritten into the dialog table. But , the sql_query can not executed. So the cdr table will not get the CDRs. I don't know why. Who can help me ? This is a simple script for testing. Any suggestion will be appreciated. Thank you in advance. ------------------ albert.gu

2 1

Issue when SRTP enabled in the SIP clients
by Jonathan Martin 21 Dec '11

21 Dec '11

Hi, I followed this web article to install Kamailio 3.2 and RTPProxy on Debian Squeeze x64: http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour The system is running on a public IP address outside of our corporate LAN. I have been testing it using pjsua v2 alpha 2 from the pjsip.org project. I am having an issue when I enable srtp in the pjsua clients. If both pjsua clients are running on machines on our corporate LAN (symmetric NAT), the call succeeds and I get audio and video. If one of the clients is running outside of the corporate LAN, the call connects but I do not get any audio or video. If I turn off srtp in both clients and try the call again, audio and video starts working. Is there any additional configuration I need to make in the kamailio.cfg file when I intend to use srtp in the clients? RTPProxy info: Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications Kamailio info: version: kamailio 3.2.0 (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 10:23:25 Nov 2 2011 with gcc 4.4.5 Regards, --Jonathan

2 1

KAMAILIO
by Ahmed Yousef 21 Dec '11

21 Dec '11

is KAMAILIO work as registrar server

2 1

回复： Why? there isn't a syslog.conf under mykamailiodirectory?
by sunny_day 21 Dec '11

21 Dec '11

Elena, your advice is very useful for me. Thank you very much. :-) However,i think i should introduce the environment I now used. First, there are 3 users on the linux, and one kamailio.cfg for each user. I am not the root user. So i don't have the priviledge to write the diretory---->/var/log/messages According the introduction you provided for me, I want to setup my own log file. But I am not sure whether it will work. The idea is: 1. In kamailio.cfg set "log_facility=LOG_LOCAL5" 2. In syslog.conf set # Log anything (except mail) of level info or higher. # Don't log private authentication messages! *.info;mail.none;authpriv.none;cron.none;local5.none /var/log/messages 3. In syslog.conf set # log messages with LOG_LOCAL5 in /var/log/kamailio.log local5.* -/home/albert/mylocal/myvar/log/kamailio.log Any idea will be appreciated. Thanks in advance. ------------------ albert.gu ------------------ 原始邮件 ------------------ 发件人: "Elena-Ramona Modroiu"<ramona(a)asipto.com>; 发送时间: 2011年12月21日(星期三) 凌晨4:29 收件人: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org>; 抄送: "sunny_day"<416415455(a)qq.com>; 主题: Re: [SR-Users] 回复： Why? there isn't a syslog.conf under mykamailiodirectory? Hi, regarding the configuration of syslog to write kamailio log messges in a special file, you can find couple of tutorials on kamailio.org wiki sites. One is: http://www.kamailio.org/wiki/tutorials/3.2.x/syslog Regards, Ramona On 12/20/11 10:35 AM, sunny_day wrote: I got it . Thank you Michal :) ------------------ albert.gu ------------------ 原始邮件 ------------------ 发件人: "Michal Matyska"<michal.matyska(a)iptel.org>; 发送时间: 2011年12月20日(星期二) 下午5:20 收件人: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - UsersMailing List"<sr-users(a)lists.sip-router.org>; 主题: Re: [SR-Users] Why? there isn't a syslog.conf under mykamailiodirectory? Hi, use -E option and you'll get all the logs on the sterr. Michal sunny_day píše v �t 20. 12. 2011 v 17:10 +0800: > Olle , Thank you very much for your reply! > I am not the root user on the unix system. but I want to customize the > log file for the function: xlog. > Now, I don't know where to see the content what xlog writes. It can > help me to debug the script of kamailio.cfg i edited. > > what i do is collect the CDRs by dialog and acc modules. Any > suggestion will be appreciated. > > Thanks a lot ! > > ------------------ > albert.gu > > > ------------------ Original ------------------ > From: "Olle E. Johansson"<oej(a)edvina.net>; > Date: Tue, Dec 20, 2011 04:57 PM > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > UsersMailing List"<sr-users(a)lists.sip-router.org>; > > Subject: Re: [SR-Users] Why? there isn't a syslog.conf under my > kamailiodirectory? > > > 20 dec 2011 kl. 05:57 skrev sunny_day: > > > There should be a syslog.conf file under the kamailio directory. > > But I haven't. > > > > I only want to see the contents generated by "xlog" > > Syslog is a daemon in your operating system. The configuration file is > propably in the /etc > directory. Try running "man syslog.conf" for more information about > that file. > > /O > > > _______________________________________________ > 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 > > _______________________________________________ > 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 _______________________________________________ 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 _______________________________________________ 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

回复： Why? there isn't a syslog.conf under mykamailiodirectory?
by sunny_day 20 Dec '11

20 Dec '11

I got it . Thank you Michal :) ------------------ albert.gu ------------------ 原始邮件 ------------------ 发件人: "Michal Matyska"<michal.matyska(a)iptel.org>; 发送时间: 2011年12月20日(星期二) 下午5:20 收件人: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - UsersMailing List"<sr-users(a)lists.sip-router.org>; 主题: Re: [SR-Users] Why? there isn't a syslog.conf under mykamailiodirectory? Hi, use -E option and you'll get all the logs on the sterr. Michal sunny_day píše v �t 20. 12. 2011 v 17:10 +0800: > Olle , Thank you very much for your reply! > I am not the root user on the unix system. but I want to customize the > log file for the function: xlog. > Now, I don't know where to see the content what xlog writes. It can > help me to debug the script of kamailio.cfg i edited. > > what i do is collect the CDRs by dialog and acc modules. Any > suggestion will be appreciated. > > Thanks a lot ! > > ------------------ > albert.gu > > > ------------------ Original ------------------ > From: "Olle E. Johansson"<oej(a)edvina.net>; > Date: Tue, Dec 20, 2011 04:57 PM > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > UsersMailing List"<sr-users(a)lists.sip-router.org>; > > Subject: Re: [SR-Users] Why? there isn't a syslog.conf under my > kamailiodirectory? > > > 20 dec 2011 kl. 05:57 skrev sunny_day: > > > There should be a syslog.conf file under the kamailio directory. > > But I haven't. > > > > I only want to see the contents generated by "xlog" > > Syslog is a daemon in your operating system. The configuration file is > propably in the /etc > directory. Try running "man syslog.conf" for more information about > that file. > > /O > > > _______________________________________________ > 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 > > _______________________________________________ > 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 _______________________________________________ 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

2 1

Database connections
by Mariana Arduini 20 Dec '11

20 Dec '11

Hello all, I'm running some performance tests and need to know how does the database connections work. Are they open when we start OpenSER and are kept open for each process or they just open before a query and closed afterwards? Does anyone know where I can find such info? Thanks in advance! Mariana.

2 2

Dispatcher module with call load balancing
by Mino Haluz 20 Dec '11

20 Dec '11

Hi, I would like to use the dispatcher module with algorithm 10 - call load balancing. But there are multiple things that are a bit unclear to me. I have multiple gateways that can serve different maximum number of calls. 1) Can I somehow set the maximum calls count for each dispatcher gateway, so that it can fairly distribute the calls? I know it is stateless module, but how could I solve this? 2) Is there any method how can I get the current number of calls for each gateway from the dispatcher module? 3) What is the expiration timer in the dispatcher module for dialogs if the dialog is canceled uncleanly ? 4) Imagine I will restart kamailio, it will lose the call count and the distribution will be incorrect. Maybe it could nice, that kamailio would offer MI command to set these values. Thanks for the responses, Mino

2 1

Re: [SR-Users] Why? there isn't a syslog.conf under my kamailiodirectory?
by sunny_day 20 Dec '11

20 Dec '11

Olle , Thank you very much for your reply! I am not the root user on the unix system. but I want to customize the log file for the function: xlog. Now, I don't know where to see the content what xlog writes. It can help me to debug the script of kamailio.cfg i edited. what i do is collect the CDRs by dialog and acc modules. Any suggestion will be appreciated. Thanks a lot ! ------------------ albert.gu ------------------ Original ------------------ From: "Olle E. Johansson"<oej(a)edvina.net>; Date: Tue, Dec 20, 2011 04:57 PM To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - UsersMailing List"<sr-users(a)lists.sip-router.org>; Subject: Re: [SR-Users] Why? there isn't a syslog.conf under my kamailiodirectory? 20 dec 2011 kl. 05:57 skrev sunny_day: > There should be a syslog.conf file under the kamailio directory. > But I haven't. > > I only want to see the contents generated by "xlog" Syslog is a daemon in your operating system. The configuration file is propably in the /etc directory. Try running "man syslog.conf" for more information about that file. /O _______________________________________________ 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

2 1

Why? there isn't a syslog.conf under my kamailio directory?
by sunny_day 20 Dec '11

20 Dec '11

There should be a syslog.conf file under the kamailio directory. But I haven't. I only want to see the contents generated by "xlog" Why? Who can help me ? Thanks a lot. albert.gu

2 1

force_send_socket and IPv6
by Andreas Granig 19 Dec '11

19 Dec '11

Hi, Is it possible to use force_send_socket() with an IPv6 address in Kamailio 3.1? I get the error ERROR: fix_actions: force_send_socket: could not resolve X:X:X:X:X:X:X:X:5060 where X:X:X:X:X:X:X:X is my IPv6 address. Another problem (not sure if it's a real problem but looks weird) is when I use X::X:X:X:X:X which is actually X:0:0:X:X:X:X:X, then when starting kamailio, the "Listening on" message on stdout expands it correctly, but the error message for force_send_socket drops a 0, so it says that it can't resolve 'X:0:X:X:X:X:X' which is not a valid IPv6 address, right? Andreas

3 17

rtpproxy and IP6 in c= line
by Andreas Granig 19 Dec '11

19 Dec '11

Hi, Is there a flag in the rtpproxy module to force a change of the IP protocol field? When a call comes from an IPv4 endpoint and you do the bridging to IPv6, then the c= and o= lines contain the IPv6 address, but the address family is still IP4 (e.g. "c=IN IP4 10.10.8.22" becomes "c=IN IP4 fe80::1"). In case of bridging from IPv6 to IPv4, the field is correctly set to IP4 though (e.g. "c=IN IP6 fe80::1" becomes "c=IN IP4 10.10.8.22"). Is there something special to be taken into consideration? I couldn't find any hints in the rtpproxy module that it tries or would try to set that correctly (kam-3.1.5). Andreas

1 2

Dispatcher does not send pings
by Mino Haluz 19 Dec '11

19 Dec '11

Hi, please, could someone specify the conditions when the dispatcher module sends the INFO pings to the gateways ? modparam("dispatcher", "flags", 2) modparam("dispatcher", "dstid_avp", "$avp(s:test)") modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_method", "INFO") modparam("dispatcher", "ds_ping_from", "sip:001122@test") And my dispatcher.list is: 1 sip:IP1:5060 1 sip:IP2:5060 Am I missing something? I dumped all the communication and the kamailio(dispatcher) does not send anything.. Thanks, Mino

2 5

using xhttp_rpc to send MESSAGE to SIP user from web server
by Krishna Kurapati 19 Dec '11

19 Dec '11

Dear list members, I have seen examples of using mi_fifo and mi_xmlrpc modules to send a MESSAGE or INVITE from a webserver. Since I am using xhttp to control presence policies from webserver, I would like to use xhttp_rpc module to send MESSAGE and INVITE from webserver. Are there any examples of using xhttp_rpc to achieve the function? Thanks Krish Kura

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