sr-users April 2015

sr-users@lists.kamailio.org

89 participants
143 discussions

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

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

Announcement: Kamailio is now systemd-rtc-server
by Alex Balashov 03 Aug '15

03 Aug '15

For immediate release: ATLANTA, GA (1 April 2015)--Evariste Systems LLC, an Atlanta-based software vendor specialising in Kamailio-based service delivery solutions for the VoIP ITSP market, is pleased to announce that it, in collaboration with Red Hat Software and Ringfree Communications, has finalised the absorption of the Kamailio SIP Server into the 'systemd' system management platform for Linux. The new component shall be called 'systemd-rtc-server', or 'Systemd Real-Time Communication Server'. Alex Balashov, principal of Evariste and leader of the tri-vendor collaboration effort, will officially announce the handover of the reigns of the Kamailio project to the personal leadership of Lennart Poettering at the upcoming Systemd Real Time Communications World conference, to be held in Berlin on 27-29 May of this year. John Knight, Director of GNOME 3 Integration and part-time usability consultant at Ringfree Communications, based in Hendersonville, North Carolina,was quick to summarise the triumphs of the long-standing integration effort. Remarked Knight: "The industry has recognised for years that a SIP proxy is a basic building block in the 'init' subsystem of any Linux host. In this age of multimedia communication with voice and video, it was a travesty that systemd handled time synchronisation, network configuration, login management, logging, and console, but not SIP message routing." Sean McCord, a veteran partner at Atlanta-based integrator CyCORE & Docker, was quick to concur: "SIP calls are much easier to troubleshoot with binary logs. Combined with packet captures of TLS-encrypted WebRTC calls, systemd-journald is the ultimate call setup troubleshooting methodology of the responsive, kinetic enterprise." To support the integration of Kamailio into the ecosystem of every major Linux distribution, Evariste has released new 'dbus_api' and 'pulseaudio' modules for the project. Balashov stated, "We fully expect to use the D-Bus API to achieve gnome-session integration with systemd-rtc-server-usrloc, but we aren't going to leave Windows users behind; KamailioSvcHost.exe will support Domain Controller policies for G.722 in Active Directory forests." Despite an aggressive delivery timeline by the tri-vendor consortium behind systemd-rtc-server, industry commentators have widely lambasted the fact that it took so long for Kamailio to become integrated into systemd. Fred Posner, solutions architect at The Palner Group in Fort Lauderdale, Florida, recently wrote in a widely-publicised blog post: "sr-dev have been keeping their heads in the sand for too long. For years now, it has been completely obvious and self-evident to anyone with half a brain that all kinds of VoIP software should be included in systemd. It's a basic building block of the whole OS, having absorbed functionality previously provided by all kinds of packages like util-linux and wireless-tools." John Knight of Ringfree accepted the criticism readily, but advocated a forward-thinking orientation focused on breaking with the uncertainty of the past: "In the absence of a SIP component for routing calls to the PSTN, some people thought, 'systemd has no clear direction apart from the whims of its developers, and is a perpetually moving goal post.' Well, a SIP server should put an end to that whole discussion; that's exactly what was missing, and now that we have systemd-rtc-server, we've eliminated all doubts about the coherence, conceptual integrity and finality of systemd." -- 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/

5 8

Interpretation of Contact header
by Joachim Büchse 30 Jul '15

30 Jul '15

Good day, I’m experiencing some problems with our VoiP providers handling of REGISTER requests. We are using a Gigaset PRO N720 as UAC behind a Juniper SSG 140 with SIP-Alg enabled. This setup kind of works with UDP but our provider wants us to use TCP. With TCP enforced incoming calls don’t work. I’ve done some wire tracing and to me it seems that the providers configuration is to blame, but then - there are many RFCs out there and many NAT and UAC bug workarounds. Anyway, I wanted to get the opinion of “the" experts about how the requests send to the UAS SHOULD be correctly interpreted. The REGISTER requests/responses look like this (outside of the firewall): Protocol TCP! client port 19091 <-> server port 5060 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.b8fc Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Authorization: Digest username="90780408050", realm="pbx.peoplefone.ch", uri="sip:pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy", response="764f371a08d258157a249f8d1b852514" Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.6bda Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp>;q=0;expires=180;received="sip:212.126.160.92:19091;transport=TCP" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 The ip:port the firewall is sending those requests from is ip 212.126.160.92 port 19091. So this does NOT match the port from the Contact header. For TCP this seems rather logical to me, as one cant be listening on a TCP port and use it for sending at the same time. The UAC closes this “register connection” with TCP FIN after the register, and so does the firewall. However unfortunately subsequent requests from the provider (ie UAS) come in on port 19091 (not port 6717 from the Contact header) and the firewall simply drops them. Observations: - the server does NOT include received=212.126.160.92 in the Via of the reponse. According to RFC3581 this is mandatory when rport is present in the request, so this is probably an error in the server. - the server does include received="sip:212.126.160.92:19091;transport=TCP” in the Contact of the response. I didnt see this in any RFC (which means nothing;-) but it could be an error. - after the client received the 200 OK it closes the TCP connection. - the server tries several times to re-contact the client (incoming TCP SYN). However not on port 6717 (defined in the Contact header) but on port 19091 (where the REGISTER came from). RFC3581 defines special behaviour when “rport” is defined in the request (i.e. response should go to the same port the request came from) - however it’s not so clear if this should apply to subsequent (INVITE/OPTIONS) requests from the server to the client. Those are strictly spoken not replies (or are they?). RFC5626 defines that a “proxy” should keep track of the flows over which it received a registration and send requests over the same flow. It is not clear if RFC5626 should be applied. The RFC5626 defines that a UAC includes an “ob” parameter in the Contact field if it whishes further requests over the same flow. Also the RFC mandates a client to add a "reg-id=x" in the Contact field. Both are not the case here, so in short I think RFC5626 should NOT be applied. In which case conecting to the originating port (instead of the Contact port) would be a server error. So in short and if I interpret the RFCs correctly, the client is reachable and should be contacted on Transport: TCP IP: 212.126.160.92 Port: 6717 If anyone who lives and breathes SIP could enlighten me if the UAS is right to call back on 19091 instead of 6717 I would really appreciate it;-)) Best regards, Joachim

3 4

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

09 Jul '15

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

11 24

kamailio crash periodically in timer handler
by Péter Barabás 27 May '15

27 May '15

Hi, we have a serious problem with Kamailio v4.2.3, it crashes after some days. We have checked the core dump and the gdb output is the following for bt and full backtrace: #0 0x0000000000602291 in timer_list_expire (t=1432559962, h=0x7f64eb5b6bd0, slow_l=0x7f64eb5ba328, slow_mark=43873) at timer.c:877 #1 0x0000000000602746 in timer_handler () at timer.c:953 #2 0x0000000000602bb9 in timer_main () at timer.c:992 #3 0x00000000004a833c in main_loop () at main.c:1700 #4 0x00000000004ad857 in main (argc=13, argv=0x7fff96697d38) at main.c:2578 #0 0x0000000000602291 in timer_list_expire (t=1432559962, h=0x7f64eb5b6bd0, slow_l=0x7f64eb5ba328, slow_mark=43873) at timer.c:877 tl = 0x7f64ebc3a908 ret = 0 #1 0x0000000000602746 in timer_handler () at timer.c:953 saved_ticks = 1432559962 run_slow_timer = 0 i = 865 __FUNCTION__ = "timer_handler" #2 0x0000000000602bb9 in timer_main () at timer.c:992 No locals. #3 0x00000000004a833c in main_loop () at main.c:1700 i = 4 pid = 0 si = 0x0 si_desc = "udp receiver child=3 sock=127.0.0.1:5060\000\375\362\250/\313\345\v\004\000\000\000\000\000\000\000\004\257\263=\000\000\000\000\200TA\000\000\000\000\000\060}i\226\377\177", '\000' <repeats 19 times>, "{i\226\377\177\000\000\376\366R\000\000\000\000\000\377\377\377\377\377\177\000\000@\350\250\000\000\000\000" nrprocs = 4 __FUNCTION__ = "main_loop" #4 0x00000000004ad857 in main (argc=13, argv=0x7fff96697d38) at main.c:2578 cfg_stream = 0x1b94010 c = -1 r = 0 tmp = 0x7fff96699b17 "" tmp_len = 0 port = 1 proto = 32767 options = 0x703e08 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 2591880768 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x0 p = 0x7fff96697c50 "" __FUNCTION__ = "main" In the logs it can be seen: Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4635]: CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 21 Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4612]: ALERT: <core> [main.c:784]: handle_sigs(): child process 4623 exited by a signal 11 Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4612]: ALERT: <core> [main.c:787]: handle_sigs(): core was generated Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4612]: INFO: <core> [main.c:799]: handle_sigs(): terminating due to SIGCHLD Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4631]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4628]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4634]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4615]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4617]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4626]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4625]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4624]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4632]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4622]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4621]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4620]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4619]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4618]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4616]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4635]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4614]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4629]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4627]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4633]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4630]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4612]: INFO: <core> [mem/f_malloc.c:599]: fm_free(): freeing a free fragment (0x7f64eca19950/0x7f64eca19988) - ignore Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4612]: INFO: <core> [mem/f_malloc.c:599]: fm_free(): freeing a free fragment (0x7f64ed587f20/0x7f64ed587f58) - ignore Kamailio runs on Ubuntu trusty. The interesting thing that we run Kamailio 4.2.3 in an other environment (on Debian Squeeze) and it works perfect, no crashes. The main difference is that in Debian environment we have less number of users and the firewall is not so strength. But the configuration and the source is the same for both. Please help us to solve this problem. Thanks. Dr. Péter Barabás Director of Software Development Arenim Technologies AB. Mob: +36 70 314 56 57 Tel: +36 1 8 555 602 E-mail: peter.barabas(a)arenim.com<mailto:peter.barabas@arenim.com> Postal: Sweden - 11435 Stockholm, Stureplan 4c, 4th. Floor [Arenim_logo]<http://www.arenim.com/> [CryptTalk_Logo]<http://www.crypttalk.com/>

2 11

NO VOICE AFTER MSILO
by sscc 19 May '15

19 May '15

i have configured msilo module successful but there isn't any voice with msilo. i debug and compare the call flow with and without msilo. with msilo in call flow it didn't follow to relay and consequently didnt activate rtpproxy befor the call is answered. due to which during call there isn't any voice. please help -- View this message in context: http://sip-router.1086192.n5.nabble.com/NO-VOICE-AFTER-MSILO-tp137430.html Sent from the Users mailing list archive at Nabble.com.

3 8

Make deb
by jay binks 07 May '15

07 May '15

is there any way to make deb packages for only 1 of the modules ?? currently it takes a long time to run "make deb" im simply trying to build and package db_cassandra ( need the package for my environment ) As a side note, is there any chance db_cassandra could be added to the kamailio deb packages ? -- Sincerely Jay

4 12

db_mongodb get columns error
by Mickael Marrache 07 May '15

07 May '15

Hi, After restarting Kamailio, I got the following errors during the startup: kamailio[12614]: ERROR: db_mongodb [mongodb_dbase.c:381]: db_mongodb_get_columns(): field [caller_sock] not found in result iterator kamailio[12614]: ERROR: db_mongodb [mongodb_dbase.c:739]: db_mongodb_store_result(): failed to set the columns kamailio[12614]: ERROR: db_mongodb [mongodb_dbase.c:918]: db_mongodb_query(): failed to store result kamailio[12614]: ERROR: db_mongodb [mongodb_dbase.c:927]: db_mongodb_query(): failed to do the query kamailio[12614]: ERROR: dialog [dlg_db_handler.c:257]: select_entire_dialog_table(): Error while querying database What may be causing the issue? I should always have a caller_sock in every dialog document. Thanks, Mickael

2 4

sht_rm_name_re() question
by Yufei Tao 05 May '15

05 May '15

Hi I've hit a problem with sht_rm_name_re() in htable module. I was calling it like this: sht_rm_name_re("Dlg=>$var(callid)::tenant"); But when I used sipp to generate 600 concurrent calls for example, I called this function when receiving BYE. But it removed more entries than it should. Seems when removing removing some entry with callid 1-3798(a)192.168.60.80, it also removed entries for [1-9]1-3798(a)192.168.60.80. How do I add '^' in front and '$' at the end of the regexp string when calling this function please? Here I don't really need regexp in fact, just want to do an exact match. But I didn't find any other functions for deleting entries. Thank you! Yufei

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sr-users April 2015