- sr-users - kamailio.org

using APP_LUA for complex arithmetic operations
by Jayesh Nambiar 03 Dec '15

03 Dec '15

Hi, I understand that kamailio does not allow float-point values in variables and I think kamailio is not a great place to do complex arithemtic operations and so I thought of using app_lua for the same. So can I pass the control to app_lua and come back to config script with some values? For eg. can I call a lua script that does a complex arithemetic operation on my AVP values and based on certain conditions; return back a true or false and then I continue in my existing script. I came across an exmaple http://kb.asipto.com/kamailio:usage:k32-lua-routing but this is more like doing the entire thing inside LUA. What I'm looking at is getting certain things done in lua and continue the script processing. Thanks for any help here. - Jayesh

3 4

Kamailio - Asterisk, problem with SIP trunk mode
by Mickael MONSIEUR 02 Dec '15

02 Dec '15

Hello, First, i have 7 years experience with Asterisk, but I started a project with Kamailio, forgive me in advance if I say silly things...! ;-) I set up a classic Asterisk / Kamailio configuration: sip phones -> kamailio -> asterisk -> sip trunks/pstn. When a call comes from the PSTN side, if I configure Asterisk as follows: [012345678] type = friend username = 012345678 secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = 012345678 Standard mode: exten => 012345678, 1, Dial(SIP/012345678) -> The call is redirected on the phone by Kamailio ! :-) ------------------------------------------------------------------------------------------------------------------------------------------------ Trunk mode: [mytrunk] type = friend username = mytrunkUser secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = mytrunkUser exten => 012345678, 1, Dial(SIP/mytrunk/012345678) -> The call is rejected by Kamailio.... exten => 012345679, 1, Dial(SIP/mytrunk/012345679) -> The call is rejected by Kamailio .... My question is how to allow the routing of multiple numbers (trunk mode) in a SIP account with Kamailio? Best regards, Mickael

3 6

PKG memory leak in 4.2.6
by Julia Boudniatsky 02 Dec '15

02 Dec '15

Hello, We are running Kamailio 4.2.6 ( db77ac) compiled with memory debugging. The free PKG memory has decreased on ~6.5 M after two weeks. Please see attached file with the log messages at kamailio shutdown. Thank you in advance, Julia.

2 5

Record-route on sequential requests
by Kristian F. Høgh 01 Dec '15

01 Dec '15

Hi list, Record-route on sequential requests doesn't have did/vsf parm. On initial INVITE, I call record_route() and set dialog flag. kamailio adds the following RR Record-Route: <sip:ww.xx.yy.zz;lr;ftag=15af612df;vsf=AAAA....;did=4a8.3ca2> UAC sends a re-INVITE, containing the following route: Route: <sip:178.21.251.54;lr;ftag=15af612df;vsf=AAAA....;did=4a8.3ca2> I call loose_route() followed by record-route(), and kamailio adds the following RR: Record-Route: <sip:ww.xx.yy.zz;lr;ftag=15af612df> I've got a phone which updates the routeset on sequential requests (which I don't think it should) and the did-matching fails. Should kamailio add did/vsf on sequential requests, when I call record_route()? Should I set dlg_match_mode to 1? (I use 0) Is it wrong to call record_route() from sequential requests? Which way is the best to resolve the problem? Regards, Kristian Høgh Uni-tel A/S

2 4

Problem with NATed UAC
by Vladimir Stepic 01 Dec '15

01 Dec '15

Hello everyone, this is my first message. I'm glad to become a part of this big community. I'm new in SER technology. I worked in Asterisk PBX only, until now. For the first time i installed Elastix MT witch have kamailio in front of asterisk, and now i have a problem. I attach a diagram of initial configuration in my lab and pcap file and kamailio.cfg. Regardless diagram, x.x.x.x addresses are public and y.y.y.y are private. When i test connection 1.1 is working, connection 1.2 not working connection 2.2 not working connection 1.3 not working connection 2.3 working Phones are connected(authenticate) to kamailio, but when i look at asterisk cli those phones are unreachable, but can call, but can not be called. On router where android works, Siemens dont, and where pc softphone works other UAC dont work. I think that my nat-traversal configure good, regardless of documentation. Please tell me, if someone has some ideas of solve this problem. Thanks in advance, and sorry for my bad english :) ---------------------------------------------------------------------- *Vladimir Stepić,* Engineer specialist in computer networking System Linux Administrator mob. tel. +381 69/18-68-431 email: e1.e398(a)gmail.com

2 1

kamailio's Makefile system : Enabling CFLAGS/LDFLAGS for udp_server.c
by aft 01 Dec '15

01 Dec '15

Hi, I'm doing some experimentation on kamailio's udp_server.c. Now i need libpcre for that. So i first thought if i enable the regex module, the CFLAGS/LDFLAGS would be available from udp_server.c. But it turns out it doesn't build(doesn't *link* to be more precise). Kamailio's build system is pretty unfamiliar for me. Although i absolutely hate autotools based build system(I prefer Scons), but i'm pretty familiar with these. But Kamailio's build system is pretty much cryptic for me(Is it based on CMake?). There is a page on devel wiki for kamailio's build system. it turns out i can do stuff like : make cfg prefix=/opt/kamailio \ SCTP=1 \ CC_EXTRA_OPTS=-I/usr/gnu/include \ group_include="standard postgres presence" \ include_modules="snmpstats lcr tls" So there might be a LD_EXTRA_OPTS or stuff like that. Which than would be defined (I guess) globally so the core c files(like udp_server.c) would be able to use that. One way i guess would be to sharpen my egrep and find that variable name. But it all feels a bit hacky. As i have already included regex module, there should be a way to reuse the CFLAGS/LDFLAGS for that module for core files like udp_server.c. Cheers Arif

2 1

Re: [SR-Users] Nathelper module
by Daniel-Constantin Mierla 01 Dec '15

01 Dec '15

Hello, On 30/11/15 14:52, Marino Mileti wrote: > > I would like to use nathelper module in order to "ping" all registered > client and "delete" (from locations table) clients that are not > reachable (maybe there's some network issue, or some other problem) > > > > I've seen that it's possible to have this feature for natted contact > using nathelper (but i haven't tried this functionaliy) > > > > Is there a way to remove records from location table without wait that > its expire is 0? > > see: http://kamailio.org/docs/modules/stable/modules/nathelper.html#nathelper.p.… Also, look at the parameter of usrloc module for handling contacts that have the tcp connection cut. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

Help Required
by anakweze charles 30 Nov '15

30 Nov '15

Hi All. Does anyone know where to find training guide for Kamailio HA? I want to have 2 Kamailio in HA with four (4) Freeswitch servers behind them. I will also like to customer cgrate to manage the accounting side in a multi-tenant environment. I am just researching for this on my own and I will appreciate a guide to realize this objective. Thanks

1 0

SNMPstats dialog 0
by Igor Potjevlesch 30 Nov '15

30 Nov '15

Hello, I have rebooted my kamailio machine and now the value of kamailio dialogs is always 0: KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInProgress.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInSetup.0 = Gauge32: 0 KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 = Counter32: 0 And also the status is: KAMAILIO-MIB::kamailioDialogUsageState.0 = INTEGER: idle(0) How can I change it to active(1)? I have dialog.so loaded as well as modparam("dialog", "dlg_flag", 4) and dlg_manage(). Thank you. Regards,

3 5

Reverted Error Message
by Sam Ware 30 Nov '15

30 Nov '15

Daniel, I got the following error message (currently running version 4.3): Nov 27 18:56:51 ip-10-1-2-100 /usr/local/sbin/kamailio[451]: ERROR: dispatcher [dispatcher.c:517]: mod_init(): call load dispatching DST ID_AVP set but no size for hash table Nov 27 18:56:51 ip-10-1-2-100 /usr/local/sbin/kamailio[451]: ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing modul e dispatcher (/usr/local/lib64/kamailio/modules/dispatcher.so) I noticed in the following DEV GIT (Version 3.1) message that the error message was changed to a message that more clearly defined the error and correction for it. But the my current message would indicate that it got changed back to the older error message. http://lists.sip-router.org/pipermail/sr-dev/2010-November/009799.html Just thought I would point it out because the newer message from the GIT message seems clearer to me. -- *Sam D Ware* CIO *Vera Networks LLC* Email: sam.ware(a)veranetworks.com

2 1

Seremis Manual
by Omnes Leroy 30 Nov '15

30 Nov '15

Hi, Is there a manual I can get so Seremis. I am a new user. Thanks, Omnes

2 1

Kamailio geolocation module
by Mititelu Stefan 28 Nov '15

28 Nov '15

Hi everyone, Is there such a thing like "geolocation" module in Kamailio? (or something related to it i.e. [1] [2]). Thanks, Stefan Mititelu [1] https://tools.ietf.org/html/rfc6442 [2] https://tools.ietf.org/html/rfc3693#page-6

2 4

crash on using the $expires variable
by Jayesh Nambiar 28 Nov '15

28 Nov '15

Hi, I am trying to extract out the expires value from the incoming REGISTER request. It might be present in the Contact header or as a separate Expires header and hence I planned to use the $expires(max) PV to get this. But kamailio just crashes as soon as I use the variable. Here's when it crashes: if(is_method("REGISTER")) { xlog("L_INFO", "Expires is $expires(max)"); if(!save("location", "0x01")) { t_reply("500", "Error while saving AOR"); } } It crashes at the xlog point. Here's the backtrace for reference: Reading symbols from /usr/local/kamailio/sbin/kamailio...done. [New LWP 16609] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid -f /usr/local/kamail'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f34b658b311 in pv_get_expires (msg=0x7f34bd879e58, param=0x7f34bd7e9220, res=0x7ffcf67af530) at pv_core.c:3060 3060 if(c->expires && c->expires->body.len) { (gdb) backtrace #0 0x00007f34b658b311 in pv_get_expires (msg=0x7f34bd879e58, param=0x7f34bd7e9220, res=0x7ffcf67af530) at pv_core.c:3060 #1 0x00000000004fb6e1 in pv_get_spec_value (msg=0x7f34bd879e58, sp=0x7f34bd7e9208, value=0x7ffcf67af530) at pvapi.c:1304 #2 0x00000000004fbb46 in pv_printf (msg=0x7f34bd879e58, list=0x7f34bd80ef38, buf=0x7f34bd8735a0 "Expires is ", len=0x7ffcf67af618) at pvapi.c:1363 #3 0x00007f34b67ea745 in xl_print_log (msg=0x7f34bd879e58, list=0x7f34bd80ef38, buf=0x7f34bd8735a0 "Expires is ", len=0x7ffcf67af618) at xl_lib.c:37 #4 0x00007f34b67eb0f9 in xlog_helper (msg=0x7f34bd879e58, xm=0x7f34bd808648, level=2, line=0, facility=-1) at xlog.c:215 #5 0x00007f34b67ec160 in xlog_2_helper (msg=0x7f34bd879e58, lev=0x7f34bd7e9130 "", frm=0x7f34bd808648 "84\177", mode=0, facility=-1) at xlog.c:284 #6 0x00007f34b67ec19c in xlog_2 (msg=0x7f34bd879e58, lev=0x7f34bd7e9130 "", frm=0x7f34bd808648 "84\177") at xlog.c:292 #7 0x000000000041e23b in do_action (h=0x7ffcf67b0b40, a=0x7f34bd817cd0, msg=0x7f34bd879e58) at action.c:1065 #8 0x000000000042a8e7 in run_actions (h=0x7ffcf67b0b40, a=0x7f34bd813e18, msg=0x7f34bd879e58) at action.c:1548 #9 0x000000000041ab16 in do_action (h=0x7ffcf67b0b40, a=0x7f34bd7e8738, msg=0x7f34bd879e58) at action.c:677 #10 0x000000000042a8e7 in run_actions (h=0x7ffcf67b0b40, a=0x7f34bd7e8738, msg=0x7f34bd879e58) at action.c:1548 #11 0x000000000041e094 in do_action (h=0x7ffcf67b0b40, a=0x7f34bd7e89c8, msg=0x7f34bd879e58) at action.c:1044 #12 0x000000000042a8e7 in run_actions (h=0x7ffcf67b0b40, a=0x7f34bd7e19d8, msg=0x7f34bd879e58) at action.c:1548 #13 0x000000000042b04d in run_top_route (a=0x7f34bd7e19d8, msg=0x7f34bd879e58, c=0x0) at action.c:1634 #14 0x000000000050f3d4 in receive_msg ( buf=0xa80c00 <buf> "REGISTER sip:devsip.tlkn.com SIP/2.0\r\nVia: SIP/2.0/UDP 203.153.53.145:8300;branch=z9hG4bK119a.e23ed10b856f38b41c93d1d259d8869f.0;i=b1\r\nVia: SIP/2.0/TCP 172.16.16.32:52358;rport=52359;received=203.153."..., len=712, rcv_info=0x7ffcf67b0e60) at receive.c:196 #15 0x0000000000613b9e in udp_rcv_loop () at udp_server.c:495 #16 0x00000000004aa119 in main_loop () at main.c:1577 #17 0x00000000004b0e1e in main (argc=5, argv=0x7ffcf67b1318) at main.c:2553 Thanks, - Jayesh

3 4

restore_uri_reply() called twice in 200OK to BYE
by Efelin Novak 27 Nov '15

27 Nov '15

Hi Folks, I have a problem with To and From headers in 200 OK to BYE request. Their URI is doubled when leaving Kamailio. Incoming 200 OK: From : <sip:00421111@IP_ADDRESS>;tag=as017a2986 To : <sip:00421222@IP_ADDRESS0>;tag=K7ryDgH2g7gFm Outgoing 200 OK: From : <sip:+421111@IP_ADDRESS><sip:+421111@IP_ADDRESS>;tag=as017a2986 To : <sip:+421222@IP_ADDRESS><sip:+421222@IP_ADDRESS>;tag=K7ryDgH2g7gFm I'm using uac module to replace these headers in auto and dialog mode: modparam("uac", "restore_mode", "auto") modparam("uac", "restore_dlg", 1) append_fromtag is also set: modparam("rr", "append_fromtag", 1) When I run kamailio in debug mode I can see the function restore_uri_reply() is called twice: . . . 0(3133) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 234, <received> = <IP_ADDRESS>; state=6 0(3133) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK0bb8c5b2>; state=6 0(3133) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 235, <rport> = <5060>; state=16 0(3133) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 0(3133) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=62 0(3133) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers(): parse_headers: this is the second via 0(3133) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=teHD72X67aQUm 0(3133) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=29 0(3133) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [60]; uri=[sip:421222@IP_ADDRESS:5060] 0(3133) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:421222@IP_ADDRESS:5060>] 0(3133) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <103> <BYE> 0(3133) DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash 10806 label 0 branch 0 0(3133) DEBUG: tm [t_lookup.c:1004]: t_reply_matching(): DEBUG: t_reply_matching: reply matched (T=0x220071b4)! 0(3133) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x220071b4, callback type 2, id 0 entered 0(3133) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=as46bccb83 0(3133) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=29 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <<sip:00421111@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <<sip:+421111@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <<sip:00421222@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <<sip:+421222@IP_ADDRESS>> 0(3133) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x220071b4, callback type 2, id 0 entered 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <<sip:00421111@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <<sip:+421111@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <<sip:00421222@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <<sip:+421222@IP_ADDRESS>> . . . I think the problem is in registering a function restore_uris_reply() in callbacks in functions (both in replace.c of uac module) * void rr_checker(struct sip_msg *msg, str *r_param, void *cb_param) * static void replace_callback(struct dlg_cell *dlg, int type, struct dlg_cb_params *_params) Both of these callbacks are registered for BYE and ACK requests (INVITE request is using different callback), therefore running restore_uri_reply() twice. For testing purposes after I have removed the callback in rr_checker and recompile the uac module, To and From headers were OK. In my configuration I'm calling uac_replace_to in RELAY route: route[RELAY] { if ( !has_totag() ) { dlg_manage(); uac_replace_to("$ru"); } if (!t_relay()) { sl_reply_error(); } } So I think there is a bug in uac module, when one function is registered twice in callbacks for NON-INVITE requests. Or is there a problem with my configuration script? Can anyone point me to the direction how to solve this issue? Thank you Efelin Kamailio version: kamailio 4.4.0-dev6 (i386/linux) a66e22-dirty

2 2

Kamailio as UAC
by Alexandru Covalschi 27 Nov '15

27 Nov '15

Hello everyone! Subj is an ethernal theme, I know, but odds are so that I just need to do that. I've configured UAC auth to successfully register and my route[PSTN] looks like that # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } if !ds_is_from_list() return; # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } remove_hf("To"); insert_hf("To: <sip:$rU@sipprovider.com>\r\n", "Call-ID"); uac_replace_from("","sip:user@sipprovider.com"); route(RELAY); exit; On INVITE's I get 407 PROXY-AUTH, which are transfered to backend FS. If I try to put if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_NOTICE", "Remote asked for authentication"); uac_auth(); } to MANAGE_FAILURE or MANAGE_REPLY route Kamailio can't start. Is that even possible? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 4

how to log sql query
by John Smolka 27 Nov '15

27 Nov '15

Hi guys, I use kamailio with mysql. Some queries are huge and when I need to debug something it takes a time, so the debbuging would be easier. Is there any way how could I log all sql queries into a log file ? Thanks, John

2 1

Media address family
by Sergey Okhapkin 27 Nov '15

27 Nov '15

What is the best way to find out address family (IPV4 or IPV6) of media in SDP? My goal is to reject INVITEs which have different address families in headers and SDP, i.e. IPv4 INVITE with IPv6 media address in SDP and vice versa.

3 2

Correct start up of RTPProxy and Kamailio
by Michael Nielsen 27 Nov '15

27 Nov '15

I've installed RTPProxy and Kamailio and would like to have them start up correct on boot. First RTPProxy and then Kamailio. How to accomplish this?

2 1

2 Kamailio servers on same domain
by Michael Nielsen 26 Nov '15

26 Nov '15

I've installed Kamailio on 2 different servers. They share the same PostgreSQL database for users etc. I've setup a single domain, sip.my-domain.com, with latency responds to the 2 servers. So a user will always land on the server with the fastest responds in a respective area of the world. However, it seems as if 2 users are registered on different Kamailio servers - even though they share the same database - they cannot call each other? Have I misunderstood something? I've enabled WITH_NAT.

2 5

kamailio very slow or stuck during period from time to time
by Thibault Gueslin 26 Nov '15

26 Nov '15

I am facing an issue with kamailio 4.3.3 that is very similar to http://lists.sip-router.org/pipermail/sr-users/2015-May/088195.html Had someone found the solution or faced similar issue ? Sometimes kamailio is receiving the message (check with wireshark, logs in tcp) but the message is not handled nor relayed. After a certain amount of time everything works fine. I will do what as has been advised : enabling pike & check with gdb when it appears. thibault

3 13

issue using sipcapture to pg database
by david escartin 26 Nov '15

26 Nov '15

hello how are you? i have a problem using sipcapture module i'm writing in postgres database some sip messages but the msg field is stored ina strange format, \x5349502f322e30203438372052657175657374205465726d696e61746 5640d0a5669613a205349502f322e302f5544502037392e3137302e36342e3136383a353038303b6272616e63683d7a39684734624b30583665375a70394e446533513b72656365697665643d37392e3137302e36342e3136383b72706f72743d353038300d0a5265636f72642d526f7574653a 203c7369703a3139302e3136362e3233322e3139333a393236323b7472616e73706f72743d7564703b6c723e2c3c7369703a37392e3137302e36382e3138363b6c723b667461673d394448397648374435444653653b6469643d3639612e616437393e0d0a43616c6c2d49443a2033323666333 037632d373138642d346164392d393234362d3333336432323466613439380d0a46726f6d3a2022313931373838393437363922203c7369703a31393137383839343736394037392e3137302e36342e3136383e3b7461673d394448397648374435444653650d0a546f3a203c7369703a313830 39323935373838374037392e3137302e36382e3138363a353036303b757365723d70686f6e653e3b7461673d736263303830367974646c746f6c302d43432d33300d0a435365713a20383338353333383820494e564954450d0a5761726e696e673a203339392031302e31382e352e363420225 353313230303031463135364c3934345b30303030305d2043616e63656c207265636569766564206f6e20696e697469616c20696e76697465220d0a436f6e74656e742d4c656e6774683a20300d0a0d0a do you know how to configure it to see it in ascii? thanks a lot and regards david escartin

2 1

[CFP] reminder! FOSDEM RTC dev-room talks: deadline Friday
by Daniel Pocock 25 Nov '15

25 Nov '15

Reminder: speaker's deadline this Friday, 27 November at 23:59 UTC We have already received several really exciting talk proposals but there is still time for people to propose talks or encourage friends or colleagues to speak. Many other dev-rooms also have a deadline in the next few days and if your topic is applicable to more than one dev-room, you are welcome to make more than one submission. Please contact us or put a note in the memo field at the top of the talk proposal if you do that. All projects are encouraged to consider making a lightning talk too, it is an excellent opportunity to get exposure for your project: even though you only have 15 minutes, it can be a much larger and more diverse audience than in some dev-rooms. For full details, please see the call for papers: http://danielpocock.com/fosdem-2016-free-rtc-dev-room-and-lounge We invite all potential speakers and participants to discuss the selection process and other aspects of FOSDEM on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc

1 0

Kamailio v4.3.4 Released
by Daniel-Constantin Mierla 25 Nov '15

25 Nov '15

Hello, Kamailio SIP Server v4.3.4 stable release is out. This is a maintenance release of the latest stable branch, 4.3, that includes fixes since release of v4.3.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.3.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.4. For more details about version 4.3.4 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/11/kamailio-v4-3-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

TLS not enough memory issue with git master
by Anthony Messina 25 Nov '15

25 Nov '15

I have noticed the following issue which began with builds somewhere between git master commits bff0a08 and 6173ef7. I did not see this issue with my previous builds and haven't been able to pin down the problem, which is why I haven't formally filed a bug. Any help or guidance is appreciated, because this has crippled my use of Kamailio. Only a restart enables it to work again until the issue recurs. ERROR: tls [tls_server.c:189]: tls_complete_init(): tls: ssl bug #1491 workaround: not enough memory for safe operation: 8870536 ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading I currently build against and run openssl-1.0.1k-12.fc22.x86_64. I have a very small operation and the only change on the operational side is that all 5 of my mobile UACs (yes, that's all) have switched from CSipSimple/Android to Zoiper/Android, which doesn't yet have support for client-side certificates so verify_certificate and require_certificate are off for both the server and client config. The server is started with: /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -m 64 -M 8 I have tried modifying the shared mem to 128 but the issue still occurs. Even right now, I am seeing the error when only one UAC has established a TLS connection: # kamcmd tls.list { id: 572 timeout: 3475 src_ip: 10.77.79.156 src_port: 58688 dst_ip: 10.77.79.3 dst_port: 5061 cipher: ECDHE-RSA-RC4-SHA SSLv3 Kx=ECDH Au=RSA Enc=RC4(128) Mac=SHA1 ct_wq_size: 0 enc_rd_buf: 0 flags: 2 state: established } # kamailio.cfg enable_tls=yes loadmodule "tls.so" modparam("tls", "connection_timeout", 60) #modparam("tls", "tls_log", 1) #modparam("tls", "tls_debug", 1) #modparam("tls", "low_mem_threshold1", -1) #modparam("tls", "low_mem_threshold2", 0) modparam("tls", "session_cache", 1) # tls.cfg [server:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/example.org.key.pem certificate = /etc/kamailio/example.org.crt.pem server_name = example.org cipher_list = ECDHE-RSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-SHA384:ECDHE-RSA- AES128-GCM-SHA256:ECDHE-RSA-AES128-SHA256:ECDHE-RSA-RC4-SHA:ECDHE-RSA-AES256- SHA:DHE-RSA-AES256-GCM-SHA384:DHE-RSA-AES256-SHA256:DHE-RSA-AES128-GCM- SHA256:DHE-RSA-AES128-SHA256:DHE-RSA-AES256-SHA:DHE-RSA-AES128-SHA:RC4- SHA:AES256-GCM-SHA384:AES256-SHA256:CAMELLIA256-SHA:ECDHE-RSA-AES128- SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:CAMELLIA128- SHA:!aNULL:!eNULL:!EXPORT:!DES:!3DES:!MD5:!PSK [client:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/example.org.key.pem certificate = /etc/kamailio/example.org.crt.pem cipher_list = ECDHE-RSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-SHA384:ECDHE-RSA- AES128-GCM-SHA256:ECDHE-RSA-AES128-SHA256:ECDHE-RSA-RC4-SHA:ECDHE-RSA-AES256- SHA:DHE-RSA-AES256-GCM-SHA384:DHE-RSA-AES256-SHA256:DHE-RSA-AES128-GCM- SHA256:DHE-RSA-AES128-SHA256:DHE-RSA-AES256-SHA:DHE-RSA-AES128-SHA:RC4- SHA:AES256-GCM-SHA384:AES256-SHA256:CAMELLIA256-SHA:ECDHE-RSA-AES128- SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:CAMELLIA128- SHA:!aNULL:!eNULL:!EXPORT:!DES:!3DES:!MD5:!PSK Thanks. -Anthony -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

3 16

Kamailio/RTPProxy incorrect port 0
by Igor Potjevlesch 25 Nov '15

25 Nov '15

Hello, I can't figure out why Kamailio reports hundreds of calls with the following error: "/usr/local/sbin/kamailio[2997]: ERROR: rtpproxy [rtpproxy.c:2727]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy" It seems that each time this error occurs, we got a one-way audio call. I already check and increase the limit for opening files that the process can open with ulimit -n 500000. But, anyway, if I look to /proc/$PID/fd, I never see more than 1000 files open. I also check the socket with netstat: same; I can see the error while there's only less than 800 sockets open. Is someone know what can be the trigger of this? Regards, Igor.

5 10

Kamailio:Contact validation during INVITE
by Prashant Desai 25 Nov '15

25 Nov '15

Hi, We observe that the following kamailio API is trying to find out user registration by comparing source IP and port with stored contact, if it fails, then it compares based on incoming SIP msg contact uri. is_registered(struct sip_msg* _m, udomain_t* _d){ getContactP(_m,_d); } pcontact_t * getContactP(struct sip_msg* _m, udomain_t* _d) {); //Getting pcontact from source IP and port if (ul.get_pcontact_by_src(_d, &received_host, _m->rcv.src_port, _m->rcv.proto, &c) == 1) { LM_DBG("No entry in usrloc for %.*s:%i (Proto %i) found!\n", received_host.len, received_host.s, _m->rcv.src_port, _m->rcv.proto); } if (c == NULL) { //Getting pcontact from contact uri. if (ul.get_pcontact(_d, &ct->uri, &c) == 0) { } } We have use case where INVITE message is from same source IP and port but having unregistered contact.With this scenario,is_registered() will return success. We would like to know reason behind comparing the source IP and port to fetch the pcontact? Regards, Prashant

3 5

Releasing Kamailio v4.3.4
by Daniel-Constantin Mierla 25 Nov '15

25 Nov '15

Hello, I am considering to release Kamailio v4.3.4 next week on Wed, Nov 25 or the following day. As usual, announce any issues that you are aware and not reported to sr-dev or bug tracker. Also, if some patch was not backported and you think it has to be, write to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

1 1

Build error on Solaris 11 with Solaris Studio
by Spencer Thomason 25 Nov '15

25 Nov '15

Hello, I realize its probably uncharted waters but I was hoping someone could point me in the right direction… I attempting to build Kamailio on a UltraSPARC T2 box running Solaris 11.3 with Sun cc and I’m not getting very far: $ CC=cc gmake all CC (cc) [kamailio] timer_proc.o cc: Warning: -xarch=v9 is deprecated, use -m64 to create 64-bit programs "timer_funcs.h", line 111: warning: argument mismatch "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "cfg/../atomic/atomic_unknown.h", line 59: #warning: no native memory barrier implementations, falling back to slow lock based workarround "cfg/../lock_alloc.h", line 72: warning: argument mismatch "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "ut.h", line 236: warning: argument mismatch "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 240: warning: argument mismatch Any help would be greatly appreciated! The contents of config.mak are below: # this file is autogenerated by make cfg MAIN_NAME= kamailio CFG_NAME= kamailio SCR_NAME= sip-router FLAVOUR= kamailio INSTALL_FLAVOUR= kamailio SRC_NAME= sip-router RELEASE= 4.3.3 OS= solaris ARCH= sparc64 C_DEFS= -DNAME='"kamailio"' -DVERSION='"4.3.3"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"cc 5.1"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=4003003 -DCFG_DIR='"/usr/local/etc/kamailio/"' -DRUN_DIR='"/var/run/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DWITH_XAVP -DDBG_QM_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DUSE_SCTP -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DUSE_PTHREAD_MUTEX -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H DEFS_RM= PROFILE= CC= cc LD= cc MKDEP= cc -xM1 MKTAGS= ctags LDFLAGS= -xO5 C_INCLUDES= CC_MKDEP_OPTS= MOD_LDFLAGS= -G -xO5 LIB_LDFLAGS= -G -xO5 UTILS_LDFLAGS= -xO5 LIB_SONAME= "-h " LD_RPATH= -"-R " LIB_SUFFIX= .so LIB_PREFIX= lib LIBS= -lfast -ldl -lresolv -L/usr/local/lib -lxnet -lnsl -lrt LEX= flex YACC= yacc YACC_FLAGS= -d -b cfg PREFIX= /usr/local LOCALBASE= /usr/local TAR= gtar INSTALL= ginstall INSTALL_CFG= ginstall -m 644 INSTALL_BIN= ginstall -m 755 INSTALL_MODULES= ginstall -m 755 INSTALL_DOC= ginstall -m 644 INSTALL_MAN= ginstall -m 644 INSTALL_LIB= ginstall -m 755 INSTALL_TOUCH= touch INSTALL_SHARE= ginstall -m 644 Q= 1 CC_EXTRA_OPTS= CPU= CFLAGS_RM= CFLAGS= -m64 -g -xO5 -fast -native -xarch=v9 -xCC -xc99 $(CC_EXTRA_OPTS) MOD_CFLAGS= -xcode=pic32 $(CFLAGS) LIB_CFLAGS= -xcode=pic32 $(CFLAGS) UTILS_CFLAGS= $(CFLAGS) BASEDIR= $(DESTDIR) basedir= $(BASEDIR) DESTDIR= LIBDIR= lib64 RUNBASEDIR= $(DESTDIR) runbasedir= $(RUNBASEDIR) PREFIX= $(LOCALBASE) prefix= $(PREFIX) cfg_prefix= $(basedir)$(prefix) cfg_dir= etc/$(MAIN_NAME)/ bin_prefix= $(basedir)$(prefix) bin_dir= sbin/ modules_prefix= $(basedir)$(prefix) modules_dir= $(LIBDIR)/$(MAIN_NAME)/ doc_prefix= $(basedir)$(prefix) doc_dir= doc/$(MAIN_NAME)/ man_prefix= $(basedir)$(prefix) man_dir= man/ ut_prefix= $(basedir)$(prefix) ut_dir= share_prefix= $(basedir)$(prefix) share_dir= share/$(MAIN_NAME)/ lib_prefix= $(basedir)$(prefix) lib_dir= $(LIBDIR)/$(MAIN_NAME)/ data_prefix= $(basedir)$(prefix) data_dir= $(MAIN_NAME)/ cfg_target= $(prefix)/$(cfg_dir) lib_target= $(prefix)/$(lib_dir) data_target= $(prefix)/$(data_dir) run_prefix= $(runbasedir) run_dir= var/run/$(MAIN_NAME)/ run_target= $(run_prefix)/$(run_dir) override makefile_defs:=1 C_DEFS:=$(filter-out $(DEFS_RM) $(extra_defs),$(C_DEFS)) $(extra_defs) CFLAGS:=$(filter-out $(CFLAGS_RM) $(CC_EXTRA_OPTS), $(CFLAGS)) $(CC_EXTRA_OPTS) Thanks! Spencer

1 0

Grab users password from WWW-Auth header
by Alexandru Covalschi 24 Nov '15

24 Nov '15

Hello! My problem is I need to do users authentication through API. So I need to replace if (!www_authenticate("$td", "subscriber")) { www_challenge("$td", "1"); } With if (!my_auth_script()) { www_challenge("$td", "1"); } The main problem is - how can I grab or compare users password? I know nonce, which I understand is MD5 salt. Can I, for example, grab users password from API, then grab the MD5 string and the nonce user sent me, calculate MD5 on base of API password and nonce - and then compare MD5 strings sent by user and calculated? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

3 11

Issue with SNMPstats
by Igor Potjevlesch 24 Nov '15

24 Nov '15

Hello, I have configured the SNMPstats module fine in my machine because I get the SNMP information with the new KAMAILIO MIBS but I don't understand why the value of kamailio dialogs is always 0: KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInProgress.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInSetup.0 = Gauge32: 0 KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 = Counter32: 0 However I can see the transactions: KAMAILIO-SIP-COMMON-MIB::kamailioSIPSummaryTotalTransactions.0 = Counter32: 54797 Should I configure an specific parametr of the SNMPstats module? Regards,

3 6

MESSAGE is not successfull in Kamailio IMS
by Thierry Luo 24 Nov '15

24 Nov '15

Hi All, I installed kamailio on Ubuntu as IMS pCSCF/iCSCF/sCSCF. REGISTER is fine, but when I tried to send a MESSAGE (SMS) from one user to another, it was always not successful. I captured IP packets on the server, and it seemed that the error was caused on pCSCF. User1 and User2 are both behind their NAT network. Suppose User1's private IP is Private_IP1, and User1's public IP (on router) is Public_IP1, and User2's private and public IPs are Private_IP2 and Public_IP2: User1 pCSCF iCSCF sCSCF user2 MESSAGE ----------------> MESSAGE ---------------------> MESSAGE -------------------> MESSAGE <----------------------------------------------------- MESSAGE ---------------------------------------------------------------------------- ----> Unfortunately, from the captured data on the server, in the last step, the MESSAGE was sent to the Private_IP2 of User2. It should be sent to the Public_IP2. Of course, the datagram with the destination of Private_IP2 cannot reach User2 successfully. Please help analyze why the pCSCF sent the MESSAGE packet to User2's private IP instead of public IP. The configuration file was attached for reference. BR. Thierry Luo

2 6

Simple User Agent change
by Steven Maddox 24 Nov '15

24 Nov '15

Hi, Sorry to ask something probably very simple but I'm difficulty finding the right kinds of resources to learn from which are succinct enough to not overload my head. We have a PBX which we don't control (and hate) which expects only certain makes/models phones to register (it checks for user agents). I have a phone which unfortunately doesn't support tweaking its user agent, so I'd like to pass it through an extremely SIP proxy to have it's user agent tweaked and all other data to/from the PBX transparently let through. I've managed to knock up something (that isn't a proxy) using Asterisk where I made a trunk and a peer... but obviously since this isn't a proxy... things like BLF and transferring calls will never work. If anyone can point me in the direction of a really simple set up to achieve this, please let me know :) -- Steven Maddox

2 2

kazoo module documentation
by Vik Killa 24 Nov '15

24 Nov '15

Hello, I'm testing kazoo module for use with RabbitMQ I noticed there is a required parameter that is not listed in the documentation: http://kamailio.org/docs/modules/devel/modules/kazoo modparam("kazoo", "pua_mode", 0) The above parameter is required but not documented. Can someone please update the documentation? Thank you. /V

2 1

Problem with resuming suspended transactions.
by Dmytro Bogovych 23 Nov '15

23 Nov '15

Greetings. We build specialized softphone for iOS. Push notifications are used to notify user and wake up softphone app on missed calls and messages. We used Daniel's example config script as base for own configuration file. It has the same parts and does not differ too much. I talk about this example: http://www.kamailio.org/events/2014-KamailioWorld/day2/26-Daniel-Constantin… Everything was ok for few monthes - with kamailio 4.1.2. Few weeks ago we upgraded to 4.3.3. Now paused calls cannot be resumed properly. Softphone receives 500 Server error (No error 0/SL) in response This is problem part of script: route[RELAYRESUMED] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. 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 ($var(rr_target) != "") { xlog("L_INFO","Relay to $var(rr_ip):$var(rr_port)\n"); $ru = "sip:" + $rU + "@" + $var(rr_ip) + ":" + $var(rr_port) + ";transport=" + $var(rr_proto); xlog("L_INFO","Current request URI is $ru\n"); if (!t_relay()) xlog("L_INFO","Cannot relay PUSHED invite."); sl_reply_error(); } else { xlog("L_WARN","Target proto is not set\n"); sl_reply_error(); } exit; } It produces log ----- Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: resuming trasaction [23927:345343979] 1985987629 (23927:345343979) Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Relay to 95.133.110.17:11791 Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Current request URI is sip:1985987629@95.133.110.17:11791;transport=tls Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No error (0/SL) Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Checking FAILED messages for 1985987629: 0 ----- sl_reply_error() sends 500 with No error (0/SL) description. Later 180 & 200 are sent. But client softphone terminates transaction already (as it handles 500 response code) . There is no "Cannot relay PUSHED invite." message in the log however. I will be happy to heary any idea what can be cause... Thank you!

2 3

Kamailio-SEMS Integration
by Prashant Desai 23 Nov '15

23 Nov '15

Hi, We are working on kamailio-4.2.3 and we require B2BUA functionality to be achieved.From kamailio official website we got to know,we can integrate kamailio with SEMS to achieve B2BUA functionality. We have few queries regarding this integration. 1.SEMS-1.6.0(latest version available) is been used,is this version suits kamailio-4.2.3? 2.SEMS-1.6.0 does not support unix based communication with kamailio-4.2.3,as SEMS-1.6.0 doesn't have unix socket interface plugin. Since we need unix socket based communication with SEMS.can we port the unix socket interface plugin to SEMS-1.6.0?If no,please suggest us better approaches. Regards, Prashant.

2 2

Kamailio, rtpproxy, websocket and 488 error
by Alfredo Pastor 23 Nov '15

23 Nov '15

Hello, I was doing some tests with Elastix MT that use version 4.1.6 of Kamailio and by default uses rtpproxy, I had the problem of error 488 and unable to make calls from websockets (Incompatible SDP). I could see that Peter Dunkley in a video explains that the problem is solved by rewriting the SDP, but he uses the mediaproxy not rtpproxy. What are your recommendations? I hope you can help me. Chreers, Alfredo.

1 0

rtpengine via-branch=extra question
by Juha Heinanen 22 Nov '15

22 Nov '15

rtpengine readme tells: via-branch=... - Include the “branch” value of one of the “Via” headers in the request to the RTP proxy. Possible values are: “1” - use the first “Via” header; “2” - use the second “Via” header; “auto” - use the first “Via” header if this is a request, or the second one if this is a reply; “extra” - don't take the value from a header, but instead use the value of the “extra_id_pv” variable. This can be used to create one media session per branch on the RTP proxy. When sending a subsequent “delete” command to the RTP proxy, you can then stop just the session for a specific branch when passing the flag '1' or '2' in the “rtpengine_delete”, or stop all sessions for a call when not passing one of those two flags there. if via-branch=extra is used in offers of branched, should it also be used in answers and deletes? -- juha

2 6

Functionality of rtpproxy
by kai.ohnacker＠cbc.de 20 Nov '15

20 Nov '15

Hello, I have installed a kamailio server with rtpproxy. The server is located between 2 networks and bridge these networks. My question: Does handle rtpproxy only 1 call at the same time? Because my configuration works, but only for one call from the first to the second network. The second call is not working. But if I make internal or external call, I can do more than one call. Do I need the multiple rtpproxy usage configuration? I thought not, because I don't use different IPs for the second call... Cheers, Kai

2 1

403 Forbidden for INVITE in kamailio for ims call
by Priyaranjan Nayak 20 Nov '15

20 Nov '15

Hi All, I have installed kamailio and using as a i-cscf,s-cscf and p-cscf server. I have taken FHoSS(OpenImscore) for hss server Now I am able to do register and unregister. When I am sending INVITE for call, the calling-party and called-party were registerd but I am getting "403 Forbidden - Dialog not found on S-CSCF or Terminating user not suitable for unregistered services". Could you please tell me what is the root cause of above error message ? Best regards, Priyaranjan

1 0

RFC: rtpproxy and a=nortpproxy:yes
by Daniel-Constantin Mierla 19 Nov '15

19 Nov '15

Hello, I am looking to get some feedback on some issues I noticed more and more lately. Apparently some SIP devices (media servers, phones, ...) are keeping the "a=nortpproxy:yes" line in SDP when replying to an INVITE that contains such line. [Alice] ------> [Kamailio+RTPProxy] ------> [Bob] The 200ok response from Bob has "a=nortpproxy:yes" in SDP. By default, that line in SDP makes the rtpproxy not to engage itself anymore in rtp relaying, and as a result things like no audio or one way audio happens. Anyone else encountering such situations? If yes, what are the devices with such behaviour? So far I noticed with some FreeSwitch and Snom -- none of them I can control, so there might be a specific configuration of those devices, not something by default there. The solution is to set: modparam("rtpproxy", "nortpproxy_str", "") and use flag 'r' for rtpproxy_manage() if the IP in SDP is not a private address. I already updated the default config for master to use flag 'r' if the SDP media IP is not private, wondering if nortpproxy_str should be set to empty in kamailio.cfg (or made empty as default in config). Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

1 0

rtpproxy database structure for kamailio 4.2.3
by Jan Hazenberg 19 Nov '15

19 Nov '15

Hi all, I'm trying to load the rtpproxy url's from database but get a error when starting kamailio (4.2.3): ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Unknown column 'set_name' in 'field list' Looks like the field set_name is missing in the database structure, but i can't find it in the docs. If i add the field kamailio starts, but what is the field set_name used for. Seems it needs to be a int value: ERROR: rtpproxy [rtpproxy.c:528]: get_rtpp_set(): Invalid set name 'test' - must be integer Thanks, Jan

2 2

Fw: Kamailio with Siremis
by Ryan Holbein 19 Nov '15

19 Nov '15

________________________________ From: Ryan Holbein <rtholbein(a)hotmail.com> Sent: Wednesday, November 18, 2015 8:07 AM To: miconda(a)gmail.com Subject: Re: [SR-Users] Kamailio with Siremis Hello, This is what I have setup so far I know it is not much but looking for help this is my very first Proxy. If you look at my Attachments I am trying to do IP auth. I have 2 Asterisk Boxs with 192.168.89.2 and .3.. The Proxy has 2 nics the outside and the 192.168.89.0/24 ... How do I setup the LCR rule if you look at the LCR Attachment? Also is my trusted permissions setup correct(Those IPs you see are correct)? If you are willing to help Thank you in Adv. ________________________________ From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Thursday, November 12, 2015 12:05 PM To: Ryan Holbein; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio with Siremis If you have a single Asterisk box, then you can simply forward using kamailio.cfg only. Look at route[PSTN] in default configuration file for an example. If you have more Asterisk boxes, then depends on what you need/want to do: - for load balancing look at dispatcher module - it has a complete sample config in the docs - for least cost routing, then lcr module is indeed an option Cheers, Daniel On 12/11/15 17:56, Ryan Holbein wrote: Now I assume in LCR I will add the Asterisk boxs so it can route? Does that sound correct? Anyone able to provide an example? Thank you ________________________________ From: sr-users <sr-users-bounces(a)lists.sip-router.org><mailto:sr-users-bounces@lists.sip-router.org> on behalf of Daniel-Constantin Mierla <miconda(a)gmail.com><mailto:miconda@gmail.com> Sent: Thursday, November 12, 2015 3:01 AM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio with Siremis Hello, look at permissions module. Default kamailio can do IP auth by enabling WITH_AUTH and WITH_IPAUTH and adding the trusted IP in database table 'address' with group id 1. Siremis has a panel for managing the records in address table. Cheers, Daniel On 11/11/15 14:10, Ryan Holbein wrote: Hello, I have setup with my phone provider an IP endpoint to do IP Auth. How would I go about setting that up in Kamailio or even better can some tell me how to do that in Siremis? _______________________________________________ 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://twitter.com/#!/miconda<http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

2 2

ndb_mongodb, db_mongodb and replicasets
by Kelvin Chua 19 Nov '15

19 Nov '15

anybody ever tried using a mongodb url with replicasets and parameters to work with kamailio? Kelvin Chua

2 11

Content-Type
by Safdar Khan 19 Nov '15

19 Nov '15

How to check content-type of ongoing various type of media streams in kamailio?

2 1

siptrace segfault - attack?
by Marian Piater 18 Nov '15

18 Nov '15

Hello, we had segfault today. Kamailio has been running for few months without problems. Nov 18 10:51:20 sbc kernel: [11326028.926502] kamailio[14452]: segfault at 30 ip 00007f20f7f3838a sp 00007ffef3ab7b10 error 4 in siptrace.so[7f20f7f23000+27000] GDB: Reading symbols from kamailio...done. [New LWP 14452] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kam'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f20f7f3838a in sip_trace (msg=0x7f2100f96750, dst=0x0, dir=0x0) at siptrace.c:1041 1041 sto.totag = get_to(msg)->tag_value; (gdb) bt #0 0x00007f20f7f3838a in sip_trace (msg=0x7f2100f96750, dst=0x0, dir=0x0) at siptrace.c:1041 #1 0x0000000000457ca9 in do_action (h=0x7ffef3ab8320, a=0x7f2100b8a928, msg=0x7f2100f96750) at action.c:1053 #2 0x0000000000463cb9 in run_actions (h=0x7ffef3ab8320, a=0x7f2100b8a928, msg=0x7f2100f96750) at action.c:1548 #3 0x00000000004643bf in run_top_route (a=0x7f2100b8a928, msg=0x7f2100f96750, c=0x0) at action.c:1634 #4 0x0000000000573ea7 in receive_msg ( buf=0x9c9400 <buf> "OPTIONS sip:100@XXX.XXX.XX.XX SIP/2.0\r\nv: SIP/2.0/UDP 69.64.39.119:5060;branch=z9hG4bK-82135822;rport\r\nContent-Length: 0\r\nf: \"MisterX\"<sip:100@1.1.1.1>;tag=61326665333131663133633401333733313630343335"..., len=357, rcv_info=0x7ffef3ab85b0) at receive.c:196 #5 0x0000000000493ff5 in udp_rcv_loop () at udp_server.c:495 #6 0x000000000051fdd7 in main_loop () at main.c:1573 #7 0x0000000000525b6b in main (argc=13, argv=0x7ffef3ab8998) at main.c:2533 (gdb) quit Our IP is hidden, but there is 69.64.39.119, which is foreign address and I think, it was attack. Unfortunatelly I don't have SIP packet details, but you can see 1.1.1.1 or strange tag or content-length=0 in received message. I looked into siptrace.c and there is function sip_trace_prepare where is get_from(msg) checked, but no get_to(msg). This function is run from main sip_trace function. So I think, we need also check get_to(msg) function. I just disabled siptrace module, but we need it. Thank you. Marian

1 1

Stop msrp relay from kamailio to freeswitch
by Safdar Khan 18 Nov '15

18 Nov '15

Hi folks I have integrated kamailio and freeswitch. The basic register and authentication is happening through kamailio and other media services through freeswitch like file sharing (using msrp). My question is how can I stop file sharing through msrp via freeswitch since everything going from kamailio to freeswitch to client. I just wanted to share file only through kamailio. I am attaching my cfg file below... #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_FREESWITCH #!define WITH_PRESENCE #!define WITH_TLS #!define WITH_MSRP #!define WITH_XHTTP #!define WITH_MSILO #!define WITH_NAT #!substdef "!MY_IP_ADDR!192.168.5.99!g" #!substdef "!MY_DOMAIN!<SERVER_IP>!g" #!substdef "!MY_WS_PORT!9080!g" #!substdef "!MY_WSS_PORT!4443!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" #!define WITH_WEBSOCKETS # Kamailio (OpenSER) SIP Server v4.3 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### 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. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!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 ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:127.0.0.1:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT listen=MY_IP_ADDR #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR #!ifdef WITH_TLS listen=MY_WSS_ADDR #!endif #!endif tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_FREESWITCH freeswitch.bindip = "192.168.5.99" desc "FreeSWITCH IP Address" freeswitch.bindport = "5090" desc "FreeSWITCH Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/local/kamailio-4.3/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.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 #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_MSRP loadmodule "msrp.so" #!endif loadmodule "rtimer.so" loadmodule "sqlops.so" loadmodule "cfgutils.so" #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" ##loadmodule "nathelper.so" #!endif #!ifdef WITH_MSILO loadmodule "msilo.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- 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 ----- # set next param to 1 to add value to ;lr param (helps with some 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) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- #modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.5.99:5060") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "NULL") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/kamailio-4.3/etc/kamailio/tls.cfg") modparam("tls", "private_key", "/usr/local/kamailio-4.3/etc/kamailio/privkey.pem") modparam("tls", "certificate", "/usr/local/kamailio-4.3/etc/kamailio/kamailio1_cert.pem") modparam("tls", "ca_list", "/usr/local/kamailio-4.3/etc/kamailio/calist.pem") modparam("tls", "verify_certificate", 1) modparam("tls", "require_certificate", 1) modparam("tls", "crl", "/usr/local/kamailio-4.3/etc/kamailio/my_crl.pem") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) #!substdef "!MSRP_MIN_EXPIRES!1800!g" #!substdef "!MSRP_MAX_EXPIRES!3600!g" # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") modparam("htable", "htable", "msrp=>size=8;autoexpire=MSRP_MAX_EXPIRES;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif #!ifdef WITH_MSRP # ----- msrp params ----- modparam("msrp", "sipmsg", 1) modparam("msrp", "cmap_size", 8) modparam("msrp", "use_path_addr", "msrp://192.168.5.99:5060") #!endif modparam("rtimer", "timer", "name=tst;interval=300;mode=1;") modparam("rtimer", "exec", "timer=tst;route=8") modparam("sqlops","sqlcon","ca=>mysql://kamailio:kamailiorw@localhost/kamailio") #!ifdef WITH_AUTH # ----- auth params ----- #modparam("auth", "nonce_count", 1) #modparam("auth", "qop", "") #!endif #!ifdef WITH_WEBSOCKETS # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is _only_ being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. #!endif #!ifdef WITH_MSILO # ------ msilo params ------ modparam("msilo","db_url",DBURL) modparam("msilo","from_address","sip:registrar@192.68.5.98") modparam("msilo","contact_hdr","Contact: <sip:registrar@192.168.5.99>\r\n") modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") modparam("msilo","offline_message","") modparam("msilo", "check_time", 200000) modparam("msilo", "expire_time", 259200) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } #!endif # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication 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 } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); #!ifdef WITH_FREESWITCH # save callee ID $avp(callee) = $rU; route(FSDISPATCH); #!endif # user location service route(LOCATION); route(RELAY); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE_MESSAGE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } if($ua =~ "friendly-scanner") { sl_send_reply("200", "OK"); exit; } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); 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()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } 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 route(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(); ## #!ifdef WITH_MSILO ### m_dump(); ### #!endif exit; } } # USER location service route[LOCATION] { xlog("going inside LOCATION\n"); #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_MSILO xlog("going inside ifdefMSILO\n"); #if(is_method("MESSAGE")) # { xlog("going inside MSILO\n"); route(MSILO); # exit; #} #!endif $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); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # 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 #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif #!ifdef WITH_MSILO if(is_method("MESSAGE")){ return; } #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) return; #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal 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("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } #!ifdef WITH_FREESWITCH # FreeSWITCH routing blocks route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } route[FSDISPATCH] { if(!is_method("INVITE")) return; if(route(FSINBOUND)) return; # dial number selection switch($rU) { case /"^41$": # 41 - voicebox menu # allow only authenticated users if($au==$null) { sl_send_reply("403", "Not allowed"); exit; } $rU = "vm-" + $au; break; case /"^441[0-9][0-9]$": # starting with 44 folowed by 1XY - direct call to voice box strip(2); route(FSVBOX); break; case /"^433[01][0-9][0-9]$": # starting with 433 folowed by (0|1)XY - conference strip(2); break; case /"^45[0-9]+$": strip(2); break; default: # offline - send to voicebox if (!registered("location")) { route(FSVBOX); exit; } # online - do bridging prefix("kb-"); if(is_method("INVITE")) { # in case of failure - re-route to FreeSWITCH VoiceMail t_on_failure("FAIL_FSVBOX"); } } route(FSRELAY); exit; } route[FSVBOX] { if(!($rU=~"^1[0-9][0-9]+$")) return; prefix("vb-"); route(FSRELAY); } # Send to FreeSWITCH route[FSRELAY] { $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":" + $sel(cfg_get.freeswitch.bindport); route(RELAY); exit; } #!endif # manage outgoing branches # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } #!ifdef WITH_MSILO #manage failure message failure_route[MANAGE_FAILURE_MESSAGE] { if (!method=="MESSAGE") { exit; } log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { t_reply("202", "Accepted"); }else{ t_reply("503", "Service Unavailable"); } } #!endif # 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")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_FREESWITCH failure_route[FAIL_FSVBOX] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { # re-route to FreeSWITCH VoiceMail $rU = $avp(callee); route(FSVBOX); } } #!endif route[8] { sql_query("ca", "insert into statistics (time_stamp,random,shm_used_size,shm_real_used_size, shm_max_used_size,shm_free_used_size,ul_users,ul_contacts) values ($Ts, $RANDOM,$stat(used_size),$stat(real_used_size),$stat(max_used_size), $stat(free_size),$stat(location-users),$stat(location-contacts))", "ra"); } event_route[msrp:frame-in] { xdbg("============#[[$msrp(method)]]===========\n"); xdbg("============*[[$si:$sp]]\n"); xdbg("============ crthop: [$msrp(crthop)]\n"); xdbg("============ prevhop: [$msrp(prevhop)]\n"); xdbg("============ nexthop: [$msrp(nexthop)]\n"); xdbg("============ firsthop: [$msrp(firsthop)]\n"); xdbg("============ lasthop: [$msrp(lasthop)]\n"); xdbg("============ prevhops: [$msrp(prevhops)]\n"); xdbg("============ nexthops: [$msrp(nexthops)]\n"); xdbg("============ srcaddr: [$msrp(srcaddr)]\n"); xdbg("============ srcsock: [$msrp(srcsock)]\n"); xdbg("============ sessid: [$msrp(sessid)]\n"); msrp_reply_flags("1"); if (msrp_is_reply()) { msrp_relay(); } else if($msrp(method)=="AUTH") { if($msrp(nexthops)>0) { msrp_relay(); exit; } # frame for local server - send Use-Path # -- passwd can be loaded from DB based on $au $var(passwd) = "xyz123"; if(!pv_www_authenticate("myrealm", "$var(passwd)", "0", "$msrp(method)")) { if(auth_get_www_authenticate("myrealm", "1", "$var(wauth)")) { msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Server Error"); } exit; } msrp_cmap_save(); } else if ($msrp(method)=="SEND" || $msrp(method)=="REPORT") { if ($msrp(nexthops)>1) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay(); exit; } if (msrp_cmap_lookup()) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay_flags("1"); msrp_relay(); } else { msrp_reply("481", "Session-does-not-exist"); } } else { msrp_reply("501", "Request-method-not-understood"); } } #--------------websockets starts here------------- #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; exit; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif #--------------websockets ends here------------- #---------------msilo starts here------------------ route[MSILO]{ #!ifdef WITH_MSILO if(!is_method("MESSAGE")) return; ##..........added...... if(lookup("location")) { if (m_store("$ru")) { if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; }; if(!lookup("location")) { $var(rc)=$rc; if (! t_newtran()) { sl_reply_error(); exit; }; switch ($var(rc)) { case -1: case -3: if (m_store("$ru")) { if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; case -2: t_on_failure("MANAGE_FAILURE_MESSAGE"); t_relay(); exit; } }; t_on_failure("MANAGE_FAILURE_MESSAGE"); t_relay(); exit; #!endif return; } #------------------msilo ends here----------------

1 0

Kamailo crashes with auth_radius module
by Volkan Oransoy 18 Nov '15

18 Nov '15

Hi, I try to enable auth_radius module with my kamailo box. But it crashes on the first REGISTER request. I have enabled it with this two lines. loadmodule "auth_radius.so" modparam("auth_radius", "radius_config", "/etc/radiusclient/radiusclient.conf") This is the authentication part of my script. if (is_method("REGISTER")) { if (!radius_www_authorize("$td")) { switch ($rc) { case -7: send_reply("500", "Server Internal Error"); exit; case -1: send_reply("400", "Bad Request"); exit; default: }; if (defined($avp(digest_challenge)) && ($avp(digest_challenge) != "")) { append_to_reply("$avp(digest_challenge)"); }; send_reply("401", "Unauthorized"); exit; }; } And in the /etc/radiusclient/dictionary file, I included sip and kamailio dictionaries. $INCLUDE dictionary.sip $INCLUDE dictionary.kamailio Service succesfully starts but it crashes on the first REGISTER request. Nov 18 11:30:27 sip kernel: [159645.360105] traps: kamailio[6941] general protection ip:7f89b1660920 sp:7ffc8d3bac78 error:0 in libc-2.19.so [7f89b15cb000+19f000] Nov 18 11:30:27 sip /usr/sbin/kamailio[6961]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 17 Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: ALERT: <core> [main.c:728]: handle_sigs(): child process 6941 exited by a signal 11 Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: ALERT: <core> [main.c:731]: handle_sigs(): core was generated Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: INFO: <core> [main.c:743]: handle_sigs(): terminating due to SIGCHLD Nov 18 11:30:27 sip /usr/sbin/kamailio[6961]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6959]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6951]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6936]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6948]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6944]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6937]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6953]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6945]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6938]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6955]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6943]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6942]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6957]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6939]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6940]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Thanks,

2 3

Authorization selects
by Alexandru Covalschi 17 Nov '15

17 Nov '15

Hello everyone! I need to extract values from authentication header, but 408. @authorization["string"] 409. @authorization["string"].username 410. @authorization["string"].username.user 411. @authorization["string"].username.domain 412. @authorization["string"].realm 413. @authorization["string"].nonce 414. @authorization["string"].uri 415. @authorization["string"].cnonce 416. @authorization["string"].nc 417. @authorization["string"].response 418. @authorization["string"].opaque 419. @authorization["string"].algorithm 420. @authorization["string"].qop these values doesn't seem to work for me. Used https://github.com/kamailio/kamailio/blob/master/doc/select_list/select_cor… doc. $var(z) = @authorization["string"].realm; xlog("Realm $var(z)"); Using that simple check I get nothing in $var(z) version: kamailio 4.3.3 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, 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 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.9.2 Can anyone check that please? Or maybe syntax is wrong and I need newer selects doc? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 3

Kamailio IMS - VM
by Bruno Cruz 17 Nov '15

17 Nov '15

Hello everyone, My name is Bruno, I'm a student from Portugal studying IMS at the moment. I've been following tutorials online, trying to get Kamailio to work as an IMS core - I'm pretty much a newbie in this field. Unfortunantly I've been getting several errors and I can't seem to properly configure everything. I was wondering if there is a All-in-One VM that can be downloaded, with Kamailio ready to be used as an IMS core? Thanks in advance. Best regards, Bruno C.

4 8

General question about mohqueue
by David Villasmil Govea 16 Nov '15

16 Nov '15

Hello list, I've been given a task to setup a predictive dialer with for 8000+ agents taking accepted calls in a queue. I've done this for a much smaller setup in freeSWITCH but i recently saw that kamailio has a mohqueue (which i already setup as a PoC) that might work for this. My question is whether you think it's a good idea to go with this on kamailio or is freeSWITCH better suited for this kind of application? The setup would be to generate outgoing calls based on a list (predictive dialer) and send answered calls to a queue to be answered by an agent. Thanks! David -- DVG -- Imagination is more important than knowledge Albert Einstein

2 1

Kamailio 4.3 and Debian 8.2 systemctl
by Grant Bagdasarian 16 Nov '15

16 Nov '15

Hello, I've just installed Kamailio 4.3 on Debian 8.2, but starting kamailio does not work since Debian requires systemctl to be setup. I've tried the below config but it didn't work (https://github.com/kamailio/kamailio/issues/294#issuecomment-138590939) usr/local/src/kamailio# systemctl status kamailio.service â kamailio.service - Kamailio SIP Server Loaded: loaded (/etc/systemd/system/kamailio.service; disabled) Active: failed (Result: resources) systemd[1]: [/etc/systemd/system/kamailio.service:14] Not an absolute path, ignoring: $PIDFILE systemd[1]: Failed to load environment files: No such file or directory systemd[1]: kamailio.service failed to run 'start' task: No such file or directory systemd[1]: Failed to start Kamailio SIP Server. systemd[1]: Unit kamailio.service entered failed state. [Unit] Description=Kamailio (OpenSER) - the Open Source SIP Server After=syslog.target network.target [Service] Type=forking Environment='PIDFILE=/var/run/kamailio/kamailio.pid' Environment='CFGFILE=/etc/kamailio/kamailio.cfg' Environment='SHM_MEMORY=64' Environment='PKG_MEMORY=8' Environment='USER=kamailio' Environment='GROUP=kamailio' EnvironmentFile=/etc/default/kamailio PIDFile=$PIDFILE # ExecStart requires a full absolute path ExecStart=/usr/sbin/kamailio -P $PIDFILE -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP ExecStopPost=/bin/rm -f $PIDFILE Restart=on-abort [Install] WantedBy=multi-user.target Does anyone have a working kamailio.service file? Regards, Grant

4 3

TLS module not initialized in 4.3.3, worked in 4.1.5
by Sebastian Damm 16 Nov '15

16 Nov '15

Hello, we just updated one kamailio server from 4.1.5 to 4.3.3, and although the config file is correct and kamailio starts up, it doesn't initialize TLS and says " tls support enabled, but no tls engine available (forgot to load the tls module?)" In the log I see: Old shutdown (last lines): Nov 13 11:44:38 lasola /usr/sbin/kamailio[15890]: DEBUG: <core> [mem/shm_mem.c:235]: shm_mem_destroy(): destroying the shared memory lock Nov 13 11:44:41 lasola /usr/sbin/kamailio[14818]: ERROR: <core> [tcp_read.c:271]: tcp_read_data(): error reading: Connection reset by peer (104) Nov 13 11:44:41 lasola /usr/sbin/kamailio[14818]: ERROR: <core> [tcp_read.c:1296]: tcp_read_req(): ERROR: tcp_read_req: error reading New startup (first lines): Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [daemonize.c:583]: set_core_dump(): core dump limits set to 18446744073709551615 Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: WARNING: <core> [main.c:2475]: main(): tls support enabled, but no tls engine available (forgot to load the tls module?) Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: WARNING: <core> [main.c:2476]: main(): disabling tls... Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [async_task.c:88]: async_task_init(): start initializing asynk task framework Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [sr_module.c:959]: init_mod(): xmlrpc Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <bind_sl> in module sl [/usr/lib/x86_64-linux-gnu/kamailio/modules//sl.so] Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [sr_module.c:959]: init_mod(): sl In our config file we have the following lines for TLS (pretty late, after all other module loading and after most parameters): #!ifdef ENABLETLS loadmodule "tls.so" modparam("tls", "private_key", "/etc/ssl/private/my.kamailio-key.pem") modparam("tls", "certificate", "/etc/ssl/certs/my.kamailio.crt") #!ifdef TLS_CA_CHAIN # Maybe we want to use a chain to the CA modparam("tls", "ca_list", "/etc/ssl/certs/my.ca-bundle.crt") #!endif enable_tls=1 listen=tls:1.2.3.4:5061 #!endif After starting up, kamailio listens on port 5060, but not on port 5061. In version 4.1.1, this config worked without a problem. Has anybody seen this before? the tls module is there and available, it doesn't say anything about "cannot load module", and it is only a warning message. I'm also wondering, why this message is the first after starting the server. From config I would expect that sl, tm and all the other modules should be initialized before tls. Best Regards, Sebastian

2 9

Detecting problems of kamailio under very high load
by aft 16 Nov '15

16 Nov '15

We have multiple kamailio servers with 4 cpu cores and 16G RAM. We use kamailio+rtpproxy as a outbound sip proxy. Usually there are many thousands of concurrent sip sessions of occurring there. Periodically sometimes it just stops serving request and spits out 5** replies. At that point we usually reload the kamailio daemon and things return back to normal. As we have a lot of servers, manually doing this is a pain in the neck. So We installed a homer/sipcapture on a separate server . And from there, we periodically scan for 500 msgs for a given time interval. If there are any, we then proceed to reload kamailio. I know its hell of an inefficient system to monitor. So I'm wondering, if there can be any more rational way to detect problems pre-emptively. any fifo/mi command which we could run to find out if the kamailio instance is hung up? Cheers Arif

3 2

Loading mohqueue stops kamailio from starting up
by David Villasmil Govea 14 Nov '15

14 Nov '15

Hello all, I installed kamailio on debian from kamailio's apt repository. Configured it with kamdbctl, started it up properly, no issues. I configured mysql also, worked properly. Next i loaded up "mohqueue" and everything went to hell :) This is my config: loadmodule "mohqueue.so" # ----- moh-queue ----- modparam ("mohqueue", "db_url", "mysql://user:password@localhost/kamailio") modparam ("mohqueue", "db_qtable", "mohqueue") modparam ("mohqueue", "db_ctable", "mohqcalls") modparam ("mohqueue", "mohdir", "/var/kamailio/MOH") and my db table for mohqueue is: +----+------------+-------------------------+--------------------+---------+-------+ | id | name | uri | mohdir | mohfile | debug | +----+------------+-------------------------+--------------------+---------+-------+ | 1 | test_queue | sip:1234@172.16.163.130 | /var/kamailio/MOH/ | test | 9 | +----+------------+-------------------------+--------------------+---------+-------+ when starting, i get: ... 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_forward_nonack> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_release> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <load_rr> in module rr [/usr/lib/x86_64-linux-gnu/kamailio/modules/rr.so] 0(5069) DEBUG: <core> [sr_module.c:695]: find_mod_export_record(): find_export_record: <rtpproxy_answer> not found 0(5069) ERROR: mohqueue [mohq.c:390]: mod_init(): Unable to load rtpproxy_answer 0(5069) ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module mohqueue (/usr/lib/x86_64-linux-gnu/kamailio/modules/mohqueue.so) ERROR: error while initializing modules Anyone can help? Thanks! -- DVG -- Imagination is more important than knowledge Albert Einstein

2 2

(no subject)
by David Villasmil Govea 14 Nov '15

14 Nov '15

Hello guys, I'm trying to startup 4.3 and i'm getting the following: 0(4420) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 192 bytes for row values at 0x7fe6bf6b5a48 0(4420) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [1] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [ sip:1234@172.16.163.130] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [/var/kamailio/MOH/] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [test.8] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [test_queue] 0(4420) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [9] 0(4420) ERROR: mohqueue [mohq_db.c:504]: update_mohq_lst(): Queue,Field (test_queue,mohdir): Unable to find MOH files (/var/kamailio/MOH//test.8.*)! 0(4420) DEBUG: <core> [db_pool.c:100]: pool_remove(): removing connection from the pool 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <bind_sl> in module sl [/usr/lib/x86_64-linux-gnu/kamailio/modules/sl.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <load_tm> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_newtran> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay_to_tcp> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay_to_udp> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_forward_nonack> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_release> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <load_rr> in module rr [/usr/lib/x86_64-linux-gnu/kamailio/modules/rr.so] 0(4420) DEBUG: <core> [sr_module.c:695]: find_mod_export_record(): find_export_record: <rtpproxy_answer> not found 0(4420) ERROR: mohqueue [mohq.c:390]: mod_init(): Unable to load rtpproxy_answer 0(4420) ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module mohqueue (/usr/lib/x86_64-linux-gnu/kamailio/modules/mohqueue.so) ERROR: error while initializing modules Any help is appreciated, Thanks! -- DVG -- Imagination is more important than knowledge Albert Einstein

2 1

Kamailio queues
by David Villasmil Govea 13 Nov '15

13 Nov '15

Hello Guys, I'm new to this list. I'm trying to setup a queue with 4.3.3 (x86_64/linux) and I see a couple of things i'm wondering about: - kamdbctl created the table "mohqueues" and the associated value in the "version" tables. Whereas kamailio uses "mohqueue" (no extra "s" at the end). - after fixing that, it finds the table, but when searching for the moh file, which i set as "/var/kamailio/MOH/test.8" on the mohqueue table, but kamailio can't find it: 0(61540) ERROR: mohqueue [mohq.c:390]: mod_init(): Unable to load rtpproxy_answer 0(61540) ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module mohqueue (/usr/lib/x86_64-linux-gnu/kamailio/modules/mohqueue.so) ERROR: error while initializing modules ls -l /var/kamailio/MOH/test.8 -rw-rw-rw- 1 root root 166038 Jun 10 2009 /var/kamailio/MOH/test.8 Can anyone lend a hand? Thanks! David -- DVG -- Imagination is more important than knowledge Albert Einstein

1 0

Re: [SR-Users] Flaky & unstable RTP connections
by Infinicalls Infinicalls 13 Nov '15

13 Nov '15

Hi, I have started RTP with # ./rtpproxy -F -l localhost -s udp:*:7890 and when I issue #netstat -pln | grep rtpp I get, udp 0 0 0.0.0.0:7890 0.0.0.0:* 8759/rtpproxy I am using another similar instance of RTP also. Still my connection seems to be flaky. This doesn't occur always. I am not using dispatcher here. Simple rtpproxy with udp:rtp1 and udp:rtp2 in kamailio.cfg file. Is there anyway to tell if RTP is healthy other than making calls and finding out? Can somebody test it from your end? Or is this problem client specific? I am using microlite and sipper on windows. phone.infinicalls.com u:test;p:test123 u:trial;p:trial123 5060 regards Ganesh Kumar

1 0

Support-for-AKAv2-MD5-authentication-in-kamailio
by Anil Kumar 13 Nov '15

13 Nov '15

Hi, I want to enable AKAv2-MD5 type authentication using a plain kamailio. Is there any way I can achieve it ? Please do needful. Thanks Anil

2 1

Kamailio with Siremis
by Ryan Holbein 12 Nov '15

12 Nov '15

Hello, I have setup with my phone provider an IP endpoint to do IP Auth. How would I go about setting that up in Kamailio or even better can some tell me how to do that in Siremis?

2 5

issue with uac_req_send to websocket peer
by Vasiliy Ganchev 12 Nov '15

12 Nov '15

Hi folk! Have a scenario: UA-A -> REGISTER -> kamailio kamailio make auth, and save(location), after this, before exit, I want to send to UA-A a custom NOTIFY with specific event (this should be done without SUBSCRIBE/PUBLISH/NOTIFY logic) .... $uac_req(method) = "NOTIFY"; $uac_req(ruri) = $sel(contact.uri); $uac_req(furi) = "sip:service@" + $rd; $uac_req(turi) = $sel(contact.uri); $uac_req(hdrs) = ......... here custom header is added xlog("L_ERR", "debuginfo, NOTIFY to ru $uac_req(ruri), fu $uac_req(furi); uac_req_send(); ............ when UA-A is connected via UDP/TCP/TLS - works OK when UA-A connected via WS/WSS", have errors: Nov 10 14:59:29 /usr/sbin/kamailio[6260]: ERROR: <script>: debuginfo, NOTIFY to ru sip:251@10.100.1.218:47114;transport=ws, fu sip:service@10.100.1.85, Nov 10 14:59:29 /usr/sbin/kamailio[6260]: WARNING: <core> [msg_translator.c:2799]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Nov 10 14:59:29 /usr/sbin/kamailio[6260]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Nov 10 14:59:29 /usr/sbin/kamailio[6260]: ERROR: tm [t_msgbuilder.c:1545]: build_uac_req(): build_uac_req(): Error while assembling Via Nov 10 14:59:29 /usr/sbin/kamailio[6260]: ERROR: tm [uac.c:379]: t_uac_prepare(): t_uac: Error while building message I see no problem with ws connect (calls to/from this ws peer works OK), immediately after registration: root@local:/home/admin# kamctl fifo ws.dump 2: wss:127.0.0.1:40239 -> wss:127.0.0.1:5061 (state: OPEN, last used 5s ago, sub-protocol: sip) The difference in UDP/TCP/TLS from WS/WSS - that first connects directly to Kamailio to port 5060/5061, but second connects to custom ports, where listen special daemon and proxy sip to kamailio 5060/5061 via localhost I tried also to add in script $uac_req(sock) = "tls:127.0.0.1:5061"; but the result is the same version: kamailio 4.2.5 (i386/linux) any suggestion? cheers -- View this message in context: http://sip-router.1086192.n5.nabble.com/issue-with-uac-req-send-to-websocke… Sent from the Users mailing list archive at Nabble.com.

2 2

Unable to find dialog
by Ryan Brindley 12 Nov '15

12 Nov '15

Hey community, Our current switch is being overloaded and when it does we're getting these 2 messages spammed in our logs: WARNING: dialog [dlg_handlers.c:1127]: dlg_onroute(): unable to find dialog for ACK with route param '35d.d2f1' [3411:7981] ERROR: tm [t_reply.c:1341]: t_should_relay_response(): ERROR: t_should_relay_response: status rewrite by UAS: stored: 503, received: 100 We're obviously working on not having the switch get so overloaded, but I'm also curious to understand why it can't find the dialogs. Is it a memory issue? Networking/dropping packets issue? I also imagine it's not as simple as a single overall answer either, but nevertheless I figured I'd reach out to the community and see if anyone has an understanding of what is happening and would be willing to take a few mins to explain. Thanks in advance, Ryan Brindley Software Development Officer Stratics Networks, Inc. 1.866.635.6918 x108

2 1

Active-Passive Auto deployment of Kamailio
by Gholamreza Sabery 11 Nov '15

11 Nov '15

Hello. Recently I started a very simple open source project for auto deploying an Active-Passive Kamailio cluster using Corosync and Pacemaker. For auto deployment I used Ansible. I thought it can be useful for some users. Link of project's repository: https://github.com/ghrst/Kamailio-HA

2 2

Kamailio can not read to_tag at BYE request
by Yuriy Gorlichenko 11 Nov '15

11 Nov '15

Hello. I have some one provider that sends me BYE request with to_tag that kamailio can not parse IP my.pro.vi.der.5060 > my.ser.vi.ce.5060: UDP, length 406 E.......x...S.0X_..L......8.BYE sip:0987654321@my.ser.vi.ce SIP/2.0 Via: SIP/2.0/UDP my.pro.vi.der:5060;branch=z9hG4bKsj7bv820c0s18k8a15q1sd000ag33.1 Call-ID: 00137fb21f65f19a3094c38b30d80f5c CSeq: 1479 BYE From: <sip:1234567890@my.pro.vi.der>;tag=SDi9r0899-5ci36vojn0 To: <sip:0987654321@my.pro.vi.der>;tag=as6635bbe1 Reason: Q.850;cause=16;text="Normal call clearing" Max-Forwards: 69 Content-Length: 0 So I guess that problem in provider (may be ";" with another code) But may be we have some instruments to get tag (without parsing strings in a config file or something like that )

2 1

Removing transport=TLS from the contact header
by Rizwan Khan 10 Nov '15

10 Nov '15

I want to remove the transport=TLS from the contact header. How can I do that inside Kamailio? The flow is an under: Client --------> Kamailio -------------> FS TLS UDP Client <-------- Kamailio <------------- FS TLS UDP FS gets a request on udp but it complains about TLS profile as the contact header has the transport=TLS. Since its receing traffic on UDP, if I only get rid of the value from the Contact Header, it would be great. Thanks, Rizwan Khan

4 15

Header field Contact not changed on kamailio 4.2
by Зуев Михаил 09 Nov '15

09 Nov '15

Hi all, I need to modify the host part of a contact header. I'm trying something like: route { ..... if (remove_hf("Contact")){ if (insert_hf("Contact: rn", "Contact")) xlog("Contact modified"); } if (!t_relay("YYY.YYY.YYY.YYY","5060")) { sl_reply_error(); } ...... } When i look SIP DUMP on YYY.YYY.YYY.YYY side i see older Contact field. I try subst_hf with regexp, subst and simply remove Contact, all this does not work. What i am doing wrong? -- С уважением Зуев Михаил

3 2

Examples
by Ryan Holbein 09 Nov '15

09 Nov '15

Hello, I have everything setup and installed... Does anyone have a good link or could tell me the steps of how to connect my trunks to phone provider and then another one would be how to route the calls to the internal PBX system. Thank you

3 4

kamailio enabling reply to options
by amar Smart Telecom 09 Nov '15

09 Nov '15

Dear All, Could you pls advise: How can we enable 'options' in Kamailio. Other server is sending "options" but our Kamailio is not responding. Thanks.

2 1

Re: [SR-Users] Branch & Forking
by Daniel-Constantin Mierla 09 Nov '15

09 Nov '15

Hello, the aggregation of replies for a parallel fork is done automatically and negative responses are absorbed. Can you send a ngrep trace of a call where you found that the negative response of a branch is not absorbed in case a branch answered? Cheers, Daniel On 05/11/15 16:41, Marino Mileti wrote: > I'm playing with branch and i'm using this simple route (copied from kamailio website) > > route { > seturi("sip:a@example.com"); > append_branch("sip:b@example.com"); > append_branch("sip:c@example.com"); > t_relay(); > break; > } > > A B & C rings at the same time. > > If B or C aren't registered, and the caller send a CANCEL, it receives also a 404 messsage because for example B wasn't online. > > Is it possible that Kamailio "absorb" this 404? I don't want that my caller receive it :) > > Marino Maria Mileti > marino.mileti(a)alice.it > > > --- > Questa e-mail è stata controllata per individuare virus con Avast antivirus. > https://www.avast.com/antivirus > > > _______________________________________________ > 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

1 0

Outgoing SIP messages
by Varghese Paul 09 Nov '15

09 Nov '15

Hi all, Is it possible to log the outgoing SIP messages relayed from kamailio ? Regards Varghese Paul

3 3

Question about siremis
by Bruno Salzano 09 Nov '15

09 Nov '15

Hello, i hope not to be OT. However I noticed that latest release is 4.2.0 and that this version is compatible with kamailio 4.2.x. Is it compatible also with 4.3.x? Another question is related to Siremis development: this great webapplication is developed using PHPOpenBiz Framework. From several days the PHPOpenBiz website is down. So will Siremis development continue on PHPOpenBiz? Is there some release schedule? Best Regards Bruno

2 1

Preprocessor define puzzle
by Alex Balashov 09 Nov '15

09 Nov '15

Hi, I've got: # define WITH_VENDOR_ROUTING_LNPOIP at the top of my config. This is intended to mean that this constant is commented out and disabled ('# define' vs. '#!define'), and tests correctly in a given request route. Later: route[STAGE_3] { ... #!ifdef WITH_VENDOR_ROUTING_LNPOIP xlog("L_INFO", "LNPOIP ENABLED STAGE 3\n"); #!else xlog("L_INFO", "LNPOIP NOT ENABLED STAGE 3\n"); #!endif route(OUTBOUND_VENDOR_SELECT); } This tests correctly: Nov 8 21:17:54 centosity6 /usr/local/sbin/kamailio[12296]: INFO: LNPOIP NOT ENABLED STAGE 3 However, in route[OUTBOUND_VENDOR_SELECT]: #!ifdef WITH_VENDOR_ROUTING_LNPOIP xlog("L_INFO", "LNPOIP ENABLED\n"); #!else xlog("L_INFO", "LNPOIP NOT ENABLED\n"); #!endif This prints: Nov 8 21:17:54 centosity6 /usr/local/sbin/kamailio[12296]: INFO: LNPOIP ENABLED Nov 8 21:17:54 centosity6 /usr/local/sbin/kamailio[12296]: INFO: LNPOIP NOT ENABLED ... which I must admit is quite puzzling. Looks like it's both defined and not defined. Running 4.3:16be09c here. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Kamailio 4.2.3 generates core dump when using sql_xquery in event_route htable:mod-init
by Jan Hazenberg 06 Nov '15

06 Nov '15

Daniel, I'm currently running a kamailio 4.2.3 with htable enabled and i'm trying to load data into a hash table using the "event_route[htable:mod-init]". When i try to execute a sql_xquery kamailio generates a core dump on startup. Example event_route: event_route[htable:mod-init] { sql_xquery("kamailio", "SELECT id,number FROM blacklist_number", "blacklist_number"); } Log output when starting kamailio: 0(16905) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(16905) INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available 0(16905) INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 0(16905) INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled 0(16905) INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: //etc/kamailio/permissions.allow 0(16905) INFO: permissions [permissions.c:608]: mod_init(): default allow file (//etc/kamailio/permissions.allow) not found => empty rule set 0(16905) INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: //etc/kamailio/permissions.deny 0(16905) INFO: permissions [permissions.c:617]: mod_init(): default deny file (//etc/kamailio/permissions.deny) not found => empty rule set 0(16905) INFO: pike [pike.c:115]: pike_init(): PIKE - initializing 0(16905) INFO: pike [ip_tree.c:88]: init_lock_set(): probing 256 set size 0(16905) INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 124928 0(16905) INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 249856 0(16905) INFO: rtpproxy [rtpproxy.c:1593]: rtpp_test(): rtp proxy <udp:10.1.8.20:22000> found, support for it enabled 0(16905) NOTICE: <script>: Startup event route! Segmentation fault (core dumped) Could this be a bug? Or is there a better way to load data from a custom table into a hashtable on startup. Thanks, Jan Hazenberg

2 2

running kamilio on mac
by Stefan Mancunianet 06 Nov '15

06 Nov '15

Hi, I’ve installed kamilio on mac, but when trying to run it i get the following error message: [sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Can sctp flag be disabled ? It’s not required for my needs, and from what I found on the web, it’s very tricky to use sctp on Mac. Thanks

3 4

Webpage Display
by Ryan Holbein 06 Nov '15

06 Nov '15

Hello, I have installed: CentOS 6 Kamailio<http://www.kamailio.org/> v4.2 Siremis v4.2 I went thought the siremis install just fine but when you get to the last page and click launch siremis the very next screen is displayed wrong. Unable to login and display any images.

3 6

segmentation fault when mongodb & tls enabled. ( kamailio 4.3.2)
by Jijo 06 Nov '15

06 Nov '15

I'm observing a segmentation fault when mongodb & tls enabled. It doesn't happen when one of the module is disabled. OS: centos 6.7 kamailio 4.3.2 mongo-c-driver version 1.1.10. It happens immediately after i register a subscriber. The location table is updated with the new data and in parallel core is generated as well. Loaded symbols for /lib64/libnss_dns-2.12.so Core was generated by `/usr/local/kamailio_proxy/sbin/kamailio -f /usr/local/kamailio_proxy/etc/kamail'. Program terminated with signal 11, Segmentation fault. #0 0x00007fd3adf65e69 in atomic_cmpxchg_int (var=0x56444e3055445030, old=0, new_v=1) at ../../atomic/atomic_x86.h:233 233 ATOMIC_FUNC_CMPXCHG(cmpxchg, "cmpxchgl %2, %1", int , int) Missing separate debuginfos, use: debuginfo-install cyrus-sasl-lib-2.1.23-15.el6_6.2.x86_64 cyrus-sasl-plain-2.1.23-15.el6_6.2.x86_64 db4-4.7.25-19.el6_6.x86_64 keyutils-libs-1.4-5.el6.x86_64 krb5-libs-1.10.3-42.el6.x86_64 libcom_err-1.41.12-22.el6.x86_64 libcurl-7.19.7-46.el6.x86_64 libidn-1.18-2.el6.x86_64 libselinux-2.0.94-5.8.el6.x86_64 libssh2-1.4.2-1.el6_6.1.x86_64 libunistring-0.9.3-5.el6.x86_64 libxml2-2.7.6-20.el6.x86_64 mysql-libs-5.1.73-5.el6_6.x86_64 nspr-4.10.8-1.el6_6.x86_64 nss-3.18.0-5.3.el6_6.x86_64 nss-util-3.18.0-1.el6_6.x86_64 openldap-2.4.40-5.el6.x86_64 openssl-1.0.1e-42.el6.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) bt #0 0x00007fd3adf65e69 in atomic_cmpxchg_int (var=0x56444e3055445030, old=0, new_v=1) at ../../atomic/atomic_x86.h:233 #1 0x00007fd3adf65eb6 in futex_get (lock=0x56444e3055445030) at ../../futexlock.h:99 #2 0x00007fd3adf66468 in prv_lock_tree_branch (b=192 '\300') at ip_tree.c:47 #3 0x00007fd3adf664e5 in lock_tree_branch (b=192 '\300') at ip_tree.c:66 #4 0x00007fd3adf6dda6 in swap_routine (ticks=123206032, param=0x0) at pike_funcs.c:293 #5 0x00000000005fef8e in compat_old_handler (ti=1971296512, tl=0x7fd32ccea430, data=0x7fd32ccea430) at timer.c:996 #6 0x00000000005ff94c in slow_timer_main () at timer.c:1130 #7 0x00000000004a8676 in main_loop () at main.c:1628 #8 0x00000000004ae38e in main (argc=11, argv=0x7fff1d97ec28) at main.c:2533 (gdb) On Clean up after the core, mongo db does segmentation fault as well Loaded symbols for /lib64/libnss_files-2.12.so Reading symbols from /lib64/libnss_dns-2.12.so...Reading symbols from /usr/lib/debug/lib64/libnss_dns-2.12.so.debug...done. done. Loaded symbols for /lib64/libnss_dns-2.12.so Core was generated by `/usr/local/kamailio_proxy/sbin/kamailio -f /usr/local/kamailio_proxy/etc/kamail'. Program terminated with signal 11, Segmentation fault. #0 __pthread_mutex_destroy (mutex=0x7fd32cce7be8) at pthread_mutex_destroy.c:28 28 if ((mutex->__data.__kind & PTHREAD_MUTEX_ROBUST_NORMAL_NP) == 0 Missing separate debuginfos, use: debuginfo-install cyrus-sasl-lib-2.1.23-15.el6_6.2.x86_64 cyrus-sasl-plain-2.1.23-15.el6_6.2.x86_64 db4-4.7.25-19.el6_6.x86_64 keyutils-libs-1.4-5.el6.x86_64 krb5-libs-1.10.3-42.el6.x86_64 libcom_err-1.41.12-22.el6.x86_64 libcurl-7.19.7-46.el6.x86_64 libidn-1.18-2.el6.x86_64 libselinux-2.0.94-5.8.el6.x86_64 libssh2-1.4.2-1.el6_6.1.x86_64 libunistring-0.9.3-5.el6.x86_64 libxml2-2.7.6-20.el6.x86_64 mysql-libs-5.1.73-5.el6_6.x86_64 nspr-4.10.8-1.el6_6.x86_64 nss-3.18.0-5.3.el6_6.x86_64 nss-util-3.18.0-1.el6_6.x86_64 openldap-2.4.40-5.el6.x86_64 openssl-1.0.1e-42.el6.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) bt #0 __pthread_mutex_destroy (mutex=0x7fd32cce7be8) at pthread_mutex_destroy.c:28 #1 0x00007fd32c86f9e6 in _mongoc_ssl_thread_cleanup () at src/mongoc/mongoc-ssl.c:555 #2 _mongoc_ssl_cleanup () at src/mongoc/mongoc-ssl.c:106 #3 0x00007fd32c8606a9 in _mongoc_do_cleanup () at src/mongoc/mongoc-init.c:127 #4 0x000000344480cdb3 in pthread_once () at ../nptl/sysdeps/unix/sysv/linux/x86_64/pthread_once.S:104 #5 0x00007fd32c8523df in __do_global_dtors_aux () from /usr/lib64/libmongoc-1.0.so.0.0.0 #6 0x0000000000000000 in ?? () (gdb) mongo-c-driver version 1.1.10 version: kamailio 4.3.2 (x86_64/linux) b5980b-dirty flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: b5980b -dirty compiled on 13:35:24 Sep 28 2015 with gcc 4.

3 11

UacRegInfo flags
by DanB 06 Nov '15

06 Nov '15

Hey Guys, I have been digging around but could not find some proper documentation on flags which are coming back in the UacRegInfo rpc query. Any chance I can find out from the flags what is the status of that registration(whether password failed or user online)? Thanks in advance for any kind of tip! DanB

2 2

Webpage Display
by Ryan Holbein 05 Nov '15

05 Nov '15

Can any help me with this .. please go to 65.61.9.54 and look at webpage how do I fix it?

1 0

q-value on unsupported clients
by Daniel W. Graham 05 Nov '15

05 Nov '15

I have a need to use q-value for serial forking but my clients don't support it. I need to specify unique q-value on all ports on the client device. I explored using the User-Agent header and appending q-value, but this wouldn't help on multi-port clients. Some things I have been thinking about are 'abusing' the authorization user or from user to somehow append q-value but this opens the door to other issues, and a database trigger however there would be no way to match this up to a record in another table as the usernames would be the same. Any thoughts/ideas or methods others have used in situations like this? -Dan

4 7

Branch & Forking
by Marino Mileti 05 Nov '15

05 Nov '15

I'm playing with branch and i'm using this simple route (copied from kamailio website) route { seturi("sip:a@example.com"); append_branch("sip:b@example.com"); append_branch("sip:c@example.com"); t_relay(); break; } A B & C rings at the same time. If B or C aren't registered, and the caller send a CANCEL, it receives also a 404 messsage because for example B wasn't online. Is it possible that Kamailio "absorb" this 404? I don't want that my caller receive it :) Marino Maria Mileti marino.mileti(a)alice.it --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

Adding proprietary event in kamailio
by Priyaranjan Nayak 05 Nov '15

05 Nov '15

Hi All, If I send a SUBSCRIBE message with "Event: presence" then I am getting 200 OK. When I am sending some proprietary event(i.e. Event: as-feature-event ) then I am getting 489 Bad event. Could you please help me how can I add proprietary event in the kamailio-4.2.5 to handle the SUBSCRIBE message with proprietary event ? Thanks Priyaranjan

2 6

pv_printf buffer for spec value
by Sven Neuhaus 05 Nov '15

05 Nov '15

Hello, I'm getting these errors since the last couple of changes I made to the Kamailio 4.3.x configuration: ERROR: <core> [pvapi.c:1361]: pv_printf(): no more space for spec value ERROR: <core> [pvapi.c:1370]: pv_printf(): buffer overflow -- increase the buffer size... I'm not even using pv_printf() so this error is a bit surprising. The first thing I tried was increasing the buffer from its default size of 1024 to 4096 (pv_buffer_size=4096) but the error messages persist. What other functions use this buffer? Regards, -Sven Neuhaus

3 10

RTP Proxy to integrate with kamailio
by Safdar Khan 05 Nov '15

05 Nov '15

Hi friends, Its been two days and still finding a way to install rtp proxy and integrate it with kamailio. I have checked running status of rtp proxy by *netstat* command in this way... *#netstat -pln | grep rttpproxy* and i got the output like that(i assume its working): *udp 0 0 127.0.0.1:7722 <http://127.0.0.1:7722> 0.0.0.0.*:* Well here is my setup of kamailio and rtp proxy... 1.I am using two computers one for running kamailio and another one for rtp proxy. 2.Both are running seperately. 3.Two android mobile phones to test the service(App using imsdroid). I am little bit confused with ip settings in kamailio configuration file in the part where *rtpproxy_sock* is given.For now i have given the *ip:port of *the computer on which rtp proxy is running separately. And i am getting the error message in *syslog* like this *can't send command to a RTP proxy* *proxy <udp:192.168.5.97:7722 <http://192.168.5.97:7722>> does not respond, > disable it* *can't get version of the RTP proxy* Actually i want to record the calls. Any kind of help to configure kamailio with rtp proxy is appreciated.

4 4

listening on multiple port
by Thibault Gueslin 04 Nov '15

04 Nov '15

I would like to set up kamailio in order to listen to multiple port. I have set : port=5060 port=5228 But it always listen on the port indicated on last line: Listening on udp: 127.0.0.1:5228 udp: x.x.x.x:5228 tcp: 127.0.0.1:5228 tcp: x.x.x.x:5228 Aliases: I don't have any "listen" in my config file. Is it supposed to work like that ?

4 7

[CFP] FOSDEM 2016, RTC devroom, speakers, volunteers neeeded
by Daniel Pocock 04 Nov '15

04 Nov '15

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2016 takes place 30-31 January 2016 in Brussels, Belgium. http://fosdem.org This email contains information about: - Real-Time communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (including the Saturday night dinner), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Saturday, 30 January 2016. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc Speaking opportunities ---------------------- Note: if you used Pentabarf before, please use the same account/username Main track: the deadline for main track presentations is midnight today, 30 October. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit Real-Time Communications dev-room: deadline 27 November Please also use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real-Time devroom". https://penta.fosdem.org/submission/FOSDEM16/ Other dev-rooms: some speakers may find their topic is in the scope of more than one dev-room. It is permitted to apply to more than one dev-room but please be kind enough to tell us if you do this. You can find the full list of dev-rooms at https://www.fosdem.org/2016/schedule/tracks/ Lightning talks: deadline 27 November The lightning talks are an excellent opportunity to introduce a wider audience to your project. Given that dev-rooms are becoming increasingly busy, all speakers are encouraged to consider applying for a lightning talk as well as a slot in the dev-room. On the "General" tab, please look for the "Track" option and choose "Lightning Talks". https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. For any questions, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 30 January - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists FOSDEM is made possible by volunteers and if you have time to contribute, please feel free to get involved through https://volunteers.fosdem.org/ Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on 28 and 29 January 2016 http://wiki.xmpp.org/web/Summit_19 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit We are also considering a more general RTC or telephony summit, potentially on 29 January. Please join the Free-RTC mailing list and send an email if you would be interested in participating, sponsoring or hosting such an event. Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 29 January On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: http://planet.jabber.org ralphm(a)ik.nu http://planet.sip5060.net daniel(a)pocock.pro http://planet.opentelecoms.org daniel(a)pocock.pro Please also link to the Planet sites from your own blog or web site. Contact ======= For discussion and queries, please join the free-rtc mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Daniel Pocock <daniel(a)pocock.pro> Ralph Meijer <ralphm(a)ik.nu> Saúl Ibarra Corretgé <saul(a)ag-projects.com> Iain R. Learmonth <irl(a)debian.org>

3 3

Re: [SR-Users] Avoid multiple registration
by Daniel-Constantin Mierla 04 Nov '15

04 Nov '15

Ok, good to know it was solved. For the records, if you want to have different number of max contact per user, then you can rely on the xavp configured to registrar xavp_cfg parameter, e.g.,: modparam("registrar", "xavp_cfg", "reg") $xavp(reg=>max_contacts) = 2; save("location"); Cheers, Daniel On 04/11/15 12:23, Marino Mileti wrote: > > Thank you Daniel!! > > I've removed the 0x04 flag and now works great!! > > > > Cheers, > > Marino > > > > On 03/11/15 17:30, Daniel-Constantin Mierla wrote: > > > > Hello, > > On 03/11/15 11:29, Marino Mileti wrote: > > I would like to avoid multiple registration with the same AOR... > > I know that there's a parameter for this (max_contacts) but if I > set it to 1, all REGISTER overwrite location info. > > > max_contacts will reject new registrations/contacts if there is > already a valid contact in the location table. > > A flag as second parameter to save(...) is the option to overwrite > existing contact with the one from current registration request. > > Cheers, > Daniel > > > > Instead i would like to reply with error if a REGISTER arrive with the > same AOR just registered on location server. > > > > Is that possible? Any help is appreciated J > > > > /Marino Maria Mileti/ > > /marino.mileti(a)alice.it <mailto:marino.mileti@alice.it>/ > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > > Avast logo <https://www.avast.com/antivirus> > > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com <https://www.avast.com/antivirus> > > > > > > _______________________________________________ > 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://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda > Book: SIP Routing With Kamailio - http://www.asipto.com > Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat > > > ------------------------------------------------------------------------ > Avast logo <https://www.avast.com/antivirus> > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com <https://www.avast.com/antivirus> > > > > > _______________________________________________ > 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

1 0

Re: [SR-Users] Avoid multiple registration
by Daniel-Constantin Mierla 04 Nov '15

04 Nov '15

Hello, On 03/11/15 11:29, Marino Mileti wrote: > > I would like to avoid multiple registration with the same AOR... > > I know that there's a parameter for this (max_contacts) but if I set > it to 1, all REGISTER overwrite location info. > max_contacts will reject new registrations/contacts if there is already a valid contact in the location table. A flag as second parameter to save(...) is the option to overwrite existing contact with the one from current registration request. Cheers, Daniel > > > > Instead i would like to reply with error if a REGISTER arrive with the > same AOR just registered on location server. > > > > Is that possible? Any help is appreciated J > > > > /Marino Maria Mileti/// > > /marino.mileti(a)alice.it <mailto:marino.mileti@alice.it>/ > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > Avast logo <https://www.avast.com/antivirus> > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com <https://www.avast.com/antivirus> > > > > > _______________________________________________ > 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

2 1

Need to increase performance to handle more SUBSCRIBE messages
by Priyaranjan Nayak 04 Nov '15

04 Nov '15

Hi All, I am sending 300 REGISTER per second and getting response properly.When I am sending 50 SUBSCRIBE per second the Kamailio is not responding properly. Now We are using kamailio-4.2.5 version. Is there any procedure to increase the performance to handle more SUBSCRIBE in a second ? Thanks, Priyaranjan

2 3

Re: [SR-Users] Avoid multiple registration
by Olle E. Johansson 04 Nov '15

04 Nov '15

> On 03 Nov 2015, at 11:29, Marino Mileti <marino.mileti(a)alice.it> wrote: > > I would like to avoid multiple registration with the same AOR... > I know that there's a parameter for this (max_contacts) but if I set it to 1, all REGISTER overwrite location info. > > Instead i would like to reply with error if a REGISTER arrive with the same AOR just registered on location server. > > Is that possible? Any help is appreciated J > A simple glance at the documentation page for the REGISTRAR shows me a function called “registred”. http://kamailio.org/docs/modules/4.3.x/modules/registrar.html#registrar.f.r… "The function returns true if the AOR in the URI is registered, false otherwise. “ If you use that one and refuse a registration if true, you will get exactly what you want. Reading available documentation is a good thing (TM) :-) Regards, /Olle

1 0

Serial Forking and Lookup Function
by Daniel W. Graham 04 Nov '15

04 Nov '15

Looking for some clarification / direction on setting up forking. I have configured clients to use q-value and is stored in location table. I see several examples as described in TM module documentation but does not mention use of lookup() function. Per registrar documentation- lookup() The functions extracts username from Request-URI and tries to find all contacts for the username in usrloc. If there are no such contacts, -1 will be returned. If there are such contacts, Request-URI will be overwritten with the contact that has the highest q value and optionally the rest will be appended to the message (depending on append_branches parameter value). Append_branch parameter The parameter controls how lookup function processes multiple contacts. If there are multiple contacts for the given username in usrloc and this parameter is set to 1, Request-URI will be overwritten with the highest-q rated contact and the rest will be appended to sip_msg structure and can be later used by tm for forking. Does lookup() build the destination set and then I can proceed by doing t_load_contacts(), t_next_contacts() and setup failure route? Any direction is appreciated! Thanks, -dan

2 2

Huge TLS performance regression migrating from 3.3.7 to 4.2.6
by Tristan Mahé 03 Nov '15

03 Nov '15

Hi List, I've been reaching a strange thing lately, trying to upgrade to kamailio 4.2.6 ( identical config ): - 35% loss of performance on TLS connections ( Sipp REGISTER scenario, easily reproduced ). - more TCP workers needed to ease the TCP queues ( 32 workers on 3.3.7, at least 64 are needed on 4.2.6 to avoid "queue full" messages in logs ). - kamailio 4.2.6 intermittent processing of requests ( Kamailio stops accepting new tcp connections, then resume a bit later, then stops again,... ) => starts happening when Kamailio is treating more than 20k tcp connections, with between 500-1k req/s. During those periods, trying to connect to kamailio with an openssl s_client -connect will end up in a timeout or with a long delay before the certificate is received ( 03 CONNECTED ). - kamailio 4.2.6 dropping connections, kamailio 3.3.7 keeping those connections: after a while, the number of active connections starts to decrease on kam4, on kam3 it stays constant. I was wondering if I was alone in this situation or if someone else is impacted, has some advices ? It does not seem to be a config issue, as it is the same that was deployed with both versions.

2 8

Testing RTP Proxy
by Safdar Khan 03 Nov '15

03 Nov '15

Hi friends, I have configured kamailio to work with rtp proxy and i have followed this link http://nil.uniza.sk/sip/kamailio/rtp-proxy-kamailio-32-configuration-debian… However i am not sure that it is working perfectly or not. My Question is.. 1.Is above link ensures that rtp proxy will work with kamailio or there is another way to deal with rtp proxy and kamailio? 2.If the above link is successful integration of rtp proxy and kamailio then how to test it. Thank you

2 1

Re: [SR-Users] Domain register_myself
by DanB 03 Nov '15

03 Nov '15

Hey Daniel, That would make sense also why it worked for me in the past! Many thanks for so fast answer! DanB

1 0

Avoid multiple registration
by Marino Mileti 03 Nov '15

03 Nov '15

I would like to avoid multiple registration with the same AOR... I know that there's a parameter for this (max_contacts) but if I set it to 1, all REGISTER overwrite location info. Instead i would like to reply with error if a REGISTER arrive with the same AOR just registered on location server. Is that possible? Any help is appreciated J Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

host for pstn.gw
by Max 03 Nov '15

03 Nov '15

Hi. I'd like to use Kamailio as a SIP proxy routing calls to upstream SIP provider. As far as I've understood the parameters I should set are: pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" The problem is that my upstream provider uses dynamic IP for its server. How can I use hostname instead of ip. >From what I see in route[PSTN] example it seems like it should just work if I use pstn.gw_ip = "sip.myprovider.lol" desc "My upstream SIP host" But if so - why the confusing name? And if not - how do I make it work? Is there pstn.gw_host or smth like that? regards, Max.

3 2

DB Problems after upgrade
by Nahum Nir 02 Nov '15

02 Nov '15

Hi All, I had a working Kamailio with Siremis. I upgraded wanted a new cfg file (big mistake, I know). I defined the DBURL at kamailio.cfg and the server start without errors, users can register but I can not make calls and Siremis missed calls table is not updated. What files and where should I update? Should I run kamdbctl? Thanks, Nir

2 1

Kamailio configuration file rtpproxy_manage not found when starting
by amar Smart Telecom 02 Nov '15

02 Nov '15

Dear All, I was trying to make such that rtpproxy is used for all calls in kamailio, Part of Kamailio.cfg looks like following: ============================= loadmodule "rtpproxy.so" # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); rtpproxy_manage("co") ; # added amar 20151015 } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } But when I run Kamailio start it gives error: Error looks like following: =================== Oct 15 16:13:42 smart kamailio[18472]: * Not starting Kamailio SIP server: invalid configuration file! Oct 15 16:13:42 smart kamailio[18472]: * Oct 15 16:13:42 smart kamailio[18472]: * 0(18480) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command rtpproxy_manage (params 1) What could be the problem?

3 3

Kamailio E164 Normalisation
by Daniel Bryars 02 Nov '15

02 Nov '15

Hello, I have a requirement to normalise a number to an international E164 inside Kamailio. In my scenario I'm receiving a call from "a platform", and then making a call out to the same platform. I'm then adding some services to that call (e.g. recording it) on the way through. The problem is that sometimes the number I receive in the FROM field in the initial invite from "A platform" has 44 at the front, but no +. "A Platform" is then rejecting the call back in from myself, because the FROM field isn't formatted correctly. Scenario 1 - WORKS FINE - sometimes we see this. PSTN -> FROM:07700900000 -> A PLATFORM -> FROM:07700900000 -> ME Call Centre <- A PLATFORM <- FROM:07700900000 <- ME Scenario 2 DOES NOT WORK - sometimes we see this. PSTN -> FROM:447700900000 -> A PLATFORM -> FROM:447700900000 -> Aeriandi Call Centre <- A PLATFORM <- FROM:447700900000 <- Aeriandi Scenario 3 WORKS by fixing up the number by adding a + PSTN -> FROM:447700900000 -> A PLATFORM -> FROM: 447700900000 -> Aeriandi Call Centre <- A PLATFORM <- FROM:+447700900000 <- Aeriandi (ADD a PLUS) So I'm investigating if it's possible to fix up the number and add the + in Kamailio. Clearly I could detect if the number starts with 44 and change this to +44, but this seems very brittle (what about other international numbers) - it there a proper way to do this, or a standard module? Unfortunately asking "A platform" to send me the "FROM" in a consistent format is not an option. Many Thanks, Daniel

3 2

kamctl fifo t_uac_dlg TLS
by Ahmed Salem 02 Nov '15

02 Nov '15

Hi, I have a kamailio tls configuration which is working fine for calls & messages In the tls configuration i set these parameters to yes for the server and client verify_certificate = yes require_certificate = yes But when I use "kamctl fifo t_uac_dlg MESSAGE" to send a message I get the following errors, when verify_certificate & require_certificate are set to no the message is sent correctly. Can somebody please help me resolve this issue. tls [tls_server.c:1193]: tls_read_f(): TLS write:error:14090086:SSL routines:SSL3_GET_SERVER_CERTIFICATE:certificate verify failed <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading tls [tls_server.c:1193]: tls_read_f(): TLS accept:error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading Thanks Ahmed

2 5

Domain register_myself
by DanB 02 Nov '15

02 Nov '15

Hey Guys, I was recently playing with "register_myself" setting of the domain module (which I was the impression that I have used it successfully in the past) and I have the feeling it does not longer behave. I have the following config: """ modparam("domain", "db_url", DBURL) modparam("domain", "register_myself", 1) .... # block foreign domains here if !is_uri_host_local() { sl_send_reply("603","Only local domains accepted"); exit; } """ Based on the register_myself setting I would expect that is_uri_host_local passes the request but it does not although the request is towards the IP address of my server. Is this the intended behavior or am I just unlucky and hit a bug? Ta, DanB

2 1

dmq_usrloc for writing to DB on one machine
by Sebastian Damm 02 Nov '15

02 Nov '15

Hi, I evaluated the dmq_usrloc module, and it works pretty good if there are two identical machines which replicate their location with each other. But what I actually want to do is to send registrations to an extra machine which is just there for writing the location into a central DB (in one table for all the registrars), from where it can be accessed (by a web page for example). So for example, I have three machines, registrar 1 and registrar 2, and receiver. receiver will never interact with customers. It is just there for writing the location into the database. I configured both registrars to send their packets to receiver, and I configured the receiver to send its packets (which it actually never does) to a dns name containing both registrars. But after reading the content of some PING messages and reading something about "autodiscovery" of dmq on the mailing list, I'm not sure whether my scenario will work the way I want it to. Can somebody explain whether I can use dmq_usrloc for this purpose? Or is it only intended to be used for bidirectional replication? Best Regards, Sebastian

2 5

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users