sr-users April 2014

sr-users@lists.kamailio.org

100 participants
171 discussions

Crash on REGISTER
by Igor Potjevlesch 03 Apr '14

03 Apr '14

Hello, I experienced an issue yesterday. I move a customer to Kamailio 4.0.4 (previously he was on an old SER instance without issue). He uses the same SIP account for connecting two IPBX (same platform, same firmware). When the second REGISTER came on Kamailio and, may be 500ms/1s after (I have sufficient time to see 200 OK (2 bindings) on Wireshark), Kamailio crash with coredump generated. I know that I must analyse the coredump but I hope that could be a known issue. In addition, when I try to run "gdb" on the core files, I got the following: Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio.pid -m 256 -M 64'. Program terminated with signal 11, Segmentation fault. #0 0x00000030f2230f30 in escape_string_for_mysql () from /usr/lib64/mysql/libmysqlclient.so.16 Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.80.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64 krb5-libs-1.9-33.el6.x86_64 libcom_err-1.41.12-12.el6.x86_64 libconfuse-2.6-2.el6.rf.x86_64 libselinux-2.0.94-5.3.el6.x86_64 mysql-libs-5.1.61-4.el6.x86_64 nss-softokn-freebl-3.12.9-11.el6.x86_64 openssl-1.0.0-20.el6_2.5.x86_64 zlib-1.2.3-27.el6.x86_64 I guess that I'm missing packages to troubleshoot. Many thanks. Regards, Igor.

3 10

NAT Traversal issue
by Wingsravi R 03 Apr '14

03 Apr '14

Dear Kamailio'ns, I am working on Kamailio server (V 4.1.2) with RTPproxy (1.2.1) integrated, in a standalone intranet infrastructure (no any connection with internet). I dont have any NAT settings in my network set-up. even though i will not get Audio/video calls through some times. So in that concern I have installed RTPproxy, now all the audio/Video calls are fine (with some Pixelled). I have the Following Kamailio configuration script, in which it suppose to invoke RTPproxy service when the SIP clients behind NAT. But every time when i do Audio/Video calls, they are proxying through RTPproxy server only. I analysed SIP captures of Audio/video call, i didnt found any IP/port changes in the whole SIP session and with this i assumed that there is no NAT issue in my Network. But why all the Audio/Video calls are proxying through RTPproxy everytime ? Is there any Wrong placement of function call in Kamailio configuration script (below) ? #----------------------------------------------------------------- #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@192.168.2.52") modparam("nathelper", "sipping_method", "INFO") # ----- NAT_traversal ----- modparam("nat_traversal", "keepalive_interval", 60) modparam("nat_traversal", "keepalive_method", "NOTIFY") modparam("nat_traversal", "keepalive_state_file", "/var/run/kamailio/keepalive_state") # ----- params needed for NAT traversal in other modules ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", 6) #!endif #Routing Script # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; #if (msg:len > max_len) { if (msg:len >= 8192 ) { sl_send_reply("513", "Message too big"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- #if (method!="REGISTER") { if (!method=="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { # unforce_rtp_proxy(); rtpproxy_answer(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("192.168.2.52","subscriber")) { # proxy_challenge("","0"); proxy_challenge("192.168.2.52", "0"); # break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; rtpproxy_offer("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- # if (uri!=myself) { if (!uri==myself) { route(4); route(1); break; }; if (method=="ACK") { route(1); break; } if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- route[1] { t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { rtpproxy_answer();; }; sl_reply_error(); }; } # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- route[2] { if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("192.168.2.52","subscriber")) { www_challenge("192.168.2.52","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- route[3] { if (!proxy_authorize("192.168.2.52","subscriber")) { proxy_challenge("192.168.2.52","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- route[4] { if (isflagset(6)) { force_rport(); fix_nated_contact(); # force_rtp_proxy(); rtpproxy_offer(); } } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { # force_rtp_proxy(); rtpproxy_offer(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } Please find the Attachment for Tcpdump based Video call Sip capture for your better Understanding. PS: Both Kamailio and RTPproxy are running on same IP (host), i.e 192.168.2.52. Please anybody help me in resolving this issue. Any help will greatly appreciate. Regards, Ravi.

3 2

when dialing another extension, instead of ringing for 30 sec, vmail kicks in right away
by mark li 03 Apr '14

03 Apr '14

I'm using kamailio as a sip proxy and registrar, and freeswitch for all media services like vmail, conference calls, ivrs etc. I just set up vmail ... and it seems to be working. Except that it appears that I've broken something else because now when I call extB from extA, instead of ringing first for a little while and then having the vmail kick in afterwards, vmail kicks in right away. someone mentioned that it could be a forking problem. I'm afraid I don't really understand what this means. just wondering if you can point me in the right direction list, as far as what I should be looking for in the kamailio.cfg file... Thanks.

2 2

error messages in kam log whenever i do anything with my phone
by mark li 02 Apr '14

02 Apr '14

Hi there. I've noticed that no matter what i do with my test phone (call voicemail, call another extension etc) I get error messages like the following in my log file: Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=553 a=26 n=xlog Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=560 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=557 a=25 n=is_method Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=563 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=560 a=25 n=is_method Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=567 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=563 a=25 n=is_method Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=570 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=567 a=24 n=t_relay Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1064 a=25 n=xdbg Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1067 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1071 a=5 n=route Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=854 a=2 n=return Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1088 a=16 n=ifApr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1073 a=24 It doesn't seem to impact the operation of the phone / dialplans etc. I recall reading somewhere that it might have to do with caller ids or location. I'm trying to google to see if I can find the article the made references to similar error messages. But in the meantime, I thought I'd check the list to see if you could point me in the right direction.

2 1

Re: [SR-Users] Regarding Websocket module
by Juha Heinanen 02 Apr '14

02 Apr '14

Premchandiran writes: > Not able to find libunistring-dev rpm for Linux 2.6.18-274.el5. almost 4 > hours I am searching for the rpm. switch to debian, -- juha

2 1

Understunding the routing logic with WebSockets
by DURECU, OLIVIER (OLIVIER) 02 Apr '14

02 Apr '14

Hi experts, Using websockets, I am facing a behavior I can't manage explaining : on reception of an initial REGISTER (over websocket), it seems that the routing logic in the request_route method is not invoked Extract of my Kamailio.cfg to explain =================================================================== Request_route { xlog("L_DBG", "Entering request_route for $rm") ... Onreply_route[REGISTER_reply] { If (t_check_status("200")) xlog("L_DBG", "REGISTER OK") Event_route[xhttp:request] { xlog("L_DBG", "XHTTP Request received") ws_handle_hanshake(); In my logs, I only see > XHTTP Request received > REGISTER OK No trace of "Entering request route for REGISTER" ! whereas for a subsequent INVITE, I see the trace Any explanation ? because of this, the fix_nated_register() is not called, and I suspect this causes the troubles I meet trying to make call with websockets Olivier DURECU Bell Labs / IP Platform Research Villarceaux Center Tel: +33(0) 130772759 GSM: +33(0) 683776482

1 0

R: Reply route and variable
by pars3c 02 Apr '14

02 Apr '14

Hello, Probably i dont see the dialog variable in the onreply route , because I see on the trace ip that sometime on the reInvite the proxy dont put on the record route the dialog cookie. On the first invite , its ok , but on every reinvite (we have sip session timer) there isnt. Its correct ? Thanks Da: pars3c [mailto:pars3c@gmail.com] Inviato: martedì 1 aprile 2014 17:32 A: 'Kamailio (SER) - Users Mailing List' Oggetto: Reply route and variable Hi , i have a question for all : in the onreply_route is visible a dialog variable ($dlg_var(ring_time)) or a avp variable ? Can I set them , on this kind of route ? On the event route (dialog start) they are all null. I try to set them, but some time (on reInvite) this variables are not setted. Why? Thanks to all

1 0

Re: [SR-Users] Regarding Websocket module
by DURECU, OLIVIER (OLIVIER) 02 Apr '14

02 Apr '14

Have a look at http://rpm.pbone.net/index.php3/stat/3/limit/1/srodzaj/1/dl/40/search/libun…<http://rpm.pbone.net/index.php3/stat/3/limit/1/srodzaj/1/dl/40/search/libun…> and search for your OS version through the pages Olivier De : sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] De la part de Premchandiran Envoyé : mercredi 2 avril 2014 09:17 À : 'Kamailio (SER) - Users Mailing List' Objet : Re: [SR-Users] Regarding Websocket module Hi Olivier, Not able to find libunistring-dev rpm for Linux 2.6.18-274.el5. almost 4 hours I am searching for the rpm. Thanks, Prem From: sr-users-bounces(a)lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org> [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of DURECU, OLIVIER (OLIVIER) Sent: Wednesday, April 02, 2014 12:39 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Regarding Websocket module Hi, You may be missing a library. On Ubunutu/debian, this is the libunistring-dev package which provides the required library This is documented on the module description (modules/websocket/README) Olivier De : sr-users-bounces(a)lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org> [mailto:sr-users-bounces@lists.sip-router.org] De la part de Premchandiran Envoyé : mercredi 2 avril 2014 09:00 À : sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> Objet : [SR-Users] Regarding Websocket module Hi All, I am using kamailio-4.1.2 and os is Linux 2.6.18-274.el5. I am getting below error though websocket.so is present in /usr/local/lib64/modules/ load_module(): ERROR: load_module: could not open module </usr/local/lib64/kamailio/modules/websocket.so>: libunistring.so.0: cannot open shared object file: No such file or directory may I know what might be the cause of this error. Regards Prem

1 0

Re: [SR-Users] Regarding Websocket module
by DURECU, OLIVIER (OLIVIER) 02 Apr '14

02 Apr '14

Hi, You may be missing a library. On Ubunutu/debian, this is the libunistring-dev package which provides the required library This is documented on the module description (modules/websocket/README) Olivier De : sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] De la part de Premchandiran Envoyé : mercredi 2 avril 2014 09:00 À : sr-users(a)lists.sip-router.org Objet : [SR-Users] Regarding Websocket module Hi All, I am using kamailio-4.1.2 and os is Linux 2.6.18-274.el5. I am getting below error though websocket.so is present in /usr/local/lib64/modules/ load_module(): ERROR: load_module: could not open module </usr/local/lib64/kamailio/modules/websocket.so>: libunistring.so.0: cannot open shared object file: No such file or directory may I know what might be the cause of this error. Regards Prem

2 1

Regarding Websocket module
by Premchandiran 02 Apr '14

02 Apr '14

Hi All, I am using kamailio-4.1.2 and os is Linux 2.6.18-274.el5. I am getting below error though websocket.so is present in /usr/local/lib64/modules/ load_module(): ERROR: load_module: could not open module </usr/local/lib64/kamailio/modules/websocket.so>: libunistring.so.0: cannot open shared object file: No such file or directory may I know what might be the cause of this error. Regards Prem

1 0

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sr-users April 2014