sr-users

sr-users@lists.kamailio.org

1 participants
37563 discussions

kamailio 4.2.1 core in timer.c:877
by 张顺通 03 Feb '15

03 Feb '15

I user rtimer module : loadmodule "rtimer.so" modparam("rtimer", "timer", "name=p_route;interval=1;mode=1;") modparam("rtimer", "exec", "timer=p_route;route=P_ROUTE") application will coredump in sometimes, and core file in below, what's the reason? Program terminated with signal 11, Segmentation fault. #0 0x00000000005fda98 in timer_list_expire (t=1043586690, h=0x7f9ed45ac9b0, slow_l=0x7f9ed45adc80, slow_mark=2316) at timer.c:877 877 timer.c: No such file or directory. in timer.c (gdb) bt full #0 0x00000000005fda98 in timer_list_expire (t=1043586690, h=0x7f9ed45ac9b0, slow_l=0x7f9ed45adc80, slow_mark=2316) at timer.c:877 tl = 0x7f9ed47f7e98 ret = 0 #1 0x00000000005fc686 in timer_handler () at timer.c:953 saved_ticks = 1043586690 run_slow_timer = 0 i = 268 __FUNCTION__ = "timer_handler" #2 0x00000000005fdd4d in timer_main () at timer.c:992 No locals. #3 0x00000000004aa571 in main_loop () at main.c:1700 i = 12 pid = 0 si = 0x0 si_desc = "udp receiver child=11 sock=10.1.*.*:5160\000\000\b\000\000\000\000\000\000\000\000\060S\324\236\177\000\000\060=#z\001\000\000\000(\205V\324\236\177\000\000\060=#z\377\177\000\000\214\216N\000\000\000\000\000\037t\372D\000\000\000\000(\205V\324\236\177\000\000\220@ #z\377\177\000\000\000\000\000\000\001\000\000" nrprocs = 12 __FUNCTION__ = "main_loop" #4 0x00000000004af166 in main (argc=11, argv=0x7fff7a234098) at main.c:2561 cfg_stream = 0x1da75e0 c = -1 r = 0 tmp = 0x7fff7a234681 "" tmp_len = 32767 port = 2049130512 proto = 32673 options = 0x706ec0 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 220012340 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xb p = 0x0 __FUNCTION__ = "main"

2 5

[OT] ssl certificates updated for kamailio.org
by Daniel-Constantin Mierla 03 Feb '15

03 Feb '15

Hello, short note to say that I updated the ssl certificates for kamailio.org web site. They are issued by cacert.org, which is probably not trusted by most of the browsers, but they are well known in the open source environment. If you get a warning about certificate change and the issuer is cacert.org, then you should be safe to login over https (e.g., to edit the wiki pages). Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Issues while using Kamailio as proxy with Asterisk
by Juned Khan 03 Feb '15

03 Feb '15

Hi All, I am using Kamailio as proxy server with Asterisk but having some issues with Inbound and outbound calls. Here are the test cases. No Test case *98 Inbound 1 ATA registered with Proxy Domain No No 2 ATA registered with Proxy IP Yes Yes 3 ATA registered with Asterisk domain No No 4 ATA registered with Asterisk IP Yes Yes 5 Soft phone registered with Proxy Domain Yes Yes 6 Soft phone registered with Proxy IP Yes Yes 7 Soft phone registered with Asterisk domain Yes Yes 8 Soft phone registered with Asterisk IP Yes Yes Here I am dialling *98 with the mentioned scenario, if it goes to voicemail options then Yes, else No I am using ALLO ATA. I have attached my kamailio.cfg with this email. Regards Juned Khan / Network Engineer +91 9974740823/ juned.khan(a)inextrix.com iNextrix Technologies Pvt Ltd. www.inextrix.com <http://htmlsig.com/www.inextrix.com> [image: Facebook] <https://www.facebook.com/junedkhan23> [image: Twitter] <https://twitter.com/juned23> [image: Google Plus] <https://plus.google.com/+JunedKhan> [image: Linkedin] <http://htmlsig.com/in.linkedin.com/in/junedk> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. E-mails are not secure and cannot be guaranteed to be error free as they can be intercepted, amended, or contain viruses. Anyone who communicates with us by e-mail is deemed to have accepted these risks. Company Name is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company.

1 0

Ro_CCR function
by Shane Harrison 03 Feb '15

03 Feb '15

Hi all, In a previous post I was alerted to the (perhaps obvious) fact that that ims_charging module is design to run within an IMS environment: http://lists.sip-router.org/pipermail/sr-users/2015-January/086675.html I have subsequently attempted to modify the module to work with the standard usrloc Kamailio module. I decided to branch from the 4.2.2 release rather than the master since I was unsure of the state of the ims_charging module in the master. It now compiles fine and installs. I also have the cdp module talking watchdog_requests to a freeDiameter server so at that level things seem OK. However inserting the Ro_CCR() call into the kamailio cfg file shows up some problems, primarily: Feb 2 01:05:52 hh-rcs-sipproxy3 /usr/sbin/kamailio[6473]: WARNING: usrloc [dlist.c:624]: register_udomain(): Registering a new domain called 'voip.we-rcs.flowcloud.systems' with usrloc Feb 2 01:05:52 hh-rcs-sipproxy3 /usr/sbin/kamailio[6473]: ERROR: <core> [db.c:443]: db_check_table_version(): invalid version 0 for table voip.we-rcs.flowcloud.systems found, expected 6 (check table structure and table "version") Exploring this error leads me to a few questions: 1) The 4.2.2 Ro_CCR() has the diameter domain as a parameter. The addition of this function to my kamailio.cfg file causes ro_fixup() -> domain_fixup() -> ul.register_domain which as per the log above, attempts to look for a table with the same name as the domain. Is it expected that such a table exists, for standard Kamailio I thought all usrloc records simply go into the Location table , regardless of the domain. 2) What is the purpose of the ro_fixup() 3) I notice that the 4.3 module documentation for ims_charging module has 6 parameters for the function call while the example code given in the same section as only 5 (missing domain). The code from the master branch for the function is different again - not sure if 4.3 is still in quite a bit of flux or not for the ims_charging module? 4) Should I be using 4.3 ims_charging module as a starting point or is 4.2.2 good enough i.e are there major changes? At this stage I am going to create a table for the domain but would like some clarity on how the usrloc and domains relate to table names. Kind regards Shane Shane Harrison Senior Software Engineer Imagination Technologies NZ Limited Level 2 1 Market Grove Lower Hutt, 5010 New Zealand PO Box 30-449 Lower Hutt, 5040 New Zealand Phone: +64 4 890-3681 ext 3361

1 2

Grouped dispatching
by Grant Bagdasarian 02 Feb '15

02 Feb '15

Hello, Would the below scenario be possible using the dispatcher module or any other routing module? Destination 1: 192.168.1.1 Destination 2: 192.168.1.2 Destination 3: 172.168.1.1 These three destinations belong to the same distribution group(set), but Kamailio has to do the following: Perform round-robin using destination 1 and 2, but if both are down fail-over to destination 3. I hope I'm being clear enough. Regards, Grant

2 1

Video Key-Frame Request using RTCP FIR or SIP INFO message
by Muhammad Shahzad 02 Feb '15

02 Feb '15

Hi, This may be a bit out of focus topic for this forum but i am posting it here anyway with hope that some guru would shed some light on it and point me to right direction. The problem is that i want to establish video call between a webrtc and a sip client using kamailio (for signalling) and RTPEngine (for media relay). Both signalling and the audio stream seems to work perfectly fine The remote video on webrtc client side (i.e. video stream from sip client) takes about 20-30 seconds to establish but once it starts it works fine. However, the remote video on sip client side (i.e. video stream from webrtc client) starts almost immediately (within 3-5 seconds) but it gets stuck after 1 or 2 seconds, then it goes blank after about 30 seconds. After a long discussion with sip client developer, we now understand the fact that sip client sends a request for so called key-frame, which is ignored by webrtc client. This request is sent through both RTCP stream and SIP INFO message. The SIP INFO message seems to be pointless as media is internally managed by chrome/firefox and these browsers don't give us such sophisticated access and control over media streams. Please let me know if this assumption is wrong. For the RTCP stream based request (RTCP-FIR), i only see "Invalid RTCP packet type" error message in RTPEngine logs (not sure if it drops this packet or relay it anyway). Does anyone has any idea on how can we either, 1. Force WebRTC client (running on Chrome / Firefox) to honor SIP INFO message and issue a key-frame in RTP video stream in response to this SIP request? OR 2. Force RTPEngine to accept RTCP-FIR and issue key-frame in RTP video stream on webrtc client's behalf? If there is any other solution to this, please feel free to share. Thank you.

4 5

What could cause an intermittent fault (28) with http_query()?
by Tim Chubb 01 Feb '15

01 Feb '15

Hi All I seem to be experiencing an intermittent fault with the utils http_query() method. We are implementing a routing and white list component that is accessed via a REST api, however i have observed several occasions where this is logged: Jan 30 16:04:17 vs-kam-prod02 /usr/local/sbin/kamailio[13184]: ERROR: utils [functions.c:149]: http_query(): failed to perform curl (28) The indicated error number (28) seems to suggest a timeout is occurring with curl, however examining a capture of network traffic when this happens shows that a http request is not sent from the server to the destination at all, usually attempting a second call results in everything working correctly. As stated its intermittent in its nature and so I cannot reliably trigger this issue, other than through sheer repetition, so any ideas as to what could be causing this issue would be gratefully received,

2 2

Async module taking down our server
by Will Ferrer 31 Jan '15

31 Jan '15

Hi All We are trying to use the async module to to delay sending a bye on from one end of the call to the other. We are using async_route(routename, seconds) to delay the WITHINDLG route. The idea is that in the future we want to be able to have our billing min duration enforced (though currently we are having issues with the dialog module that we are discussing in another thread). After running this on our deploy servers, the delays before sending on the byes get longer and longer, and then kamailio goes down. Then the receive udp buffer fills up. We tried it with both 4 and 400 async workers, and it made no difference. I am including a screen capture of the servers stats when this happens taken from voip monitor. Here are the relevant parts of the config: ... loadmodule "async.so" ... modparam("async", "workers", ASYNC_THREADS) ... request_route { ... route(DELAYED_BYE_STATIC); ... route[DELAYED_BYE_STATIC] { xlog("L_DEBUG","route DELAYED_BYE_STATIC"); #!ifdef WITH_DELAYED_BYE_STATIC if (is_method("BYE")) { xlog("L_DEBUG","route DELAYED_BYE_STATIC, from self \n"); #if (from_uri == myself) { if ((allow_trusted() || allow_source_address()) && from_uri == myself) { xlog("L_DEBUG","route DELAYED_BYE_STATIC, Bye detected, from self \n"); send_reply("200", "OK"); xlog("L_DEBUG","route DELAYED_BYE_STATIC, sent 200 about to sleep \n"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails if (has_totag()) { xlog("L_DEBUG","route DELAYED_BYE_STATIC, sleeping to WITHINDLG_DELAYED \n"); async_route("WITHINDLG_DELAYED", MIN_DURATION); } else { xlog("L_DEBUG","route DELAYED_BYE_STATIC, sleeping to WITHINDLG \n"); async_route("WITHINDLG", MIN_DURATION); } xlog("L_DEBUG","route DELAYED_BYE_STATIC, slept\n"); exit; } } #!endif return; } ... route[WITHINDLG_DELAYED] { xlog("L_DEBUG", "route WITHINDLG_DELAYED: triggered \n"); $avp(was_delayed) = 1; route(WITHINDLG); } ... route[WITHINDLG] { xlog("L_DEBUG", "route WITHINDLG: will -- DLG triggered, request method: $rm \n"); #!ifdef WITH_DISPATCHER if(is_method("BYE|CANCEL")) { xlog("L_DEBUG","route WITHINDLG: cancel or bye detected, request method: $rm \n"); #!ifdef WITH_DISPATCHER_LOAD_AWARE xlog("L_DEBUG","route WITHINDLG: running ds_load_update, request method: $rm \n"); ds_load_update(); #dlg_get ("$ci","$ft","$tt"); #dlg_bye ("all"); #!endif } #!endif if (has_totag() || $avp(was_delayed) == 1) { xlog("L_DEBUG", "route WITHINDLG: will -- DLG has totag or was_delayed: $avp(was_delayed) \n"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_DEBUG", "route WITHINDLG: will -- DLG has loose route \n"); route(DLGURI); if (is_method("BYE")) { xlog("L_DEBUG","route WITHINDLG: BYE detected"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails xlog("L_DEBUG","route WITHINDLG: ACC flag set"); } 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(); } xlog("L_DEBUG", "route WITHINDLG: will -- DLG RELAY 1\n"); route(RELAY); } else { xlog("L_DEBUG", "route WITHINDLG: will -- DLG else \n"); if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { xlog("L_DEBUG", "route WITHINDLG: will -- DLG is ack \n"); if ( t_check_trans() ) { xlog("L_DEBUG", "route WITHINDLG: will -- DLG t_check_trans \n"); # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server xlog("L_DEBUG", "route WITHINDLG: will -- DLG RELAY 2\n"); route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } ... Does any one know if this is a bug or a leak with in the async module, or perhaps something I am doing in my config? Thanks in advance for an assistance you can offer me. All the best. Will Ferrer Switchsoft

3 5

Dispatcher weight dont work
by Yuriy Gorlichenko 30 Jan '15

30 Jan '15

Hello I use dipatcher algorithm 8 that works with weight. I added 2 Asterisks and try to call its with my kam.We use 4.3 version. Tthis config select needed dst from database with my scenario. if(!ds_select_dst("$var(setid)", "8")) $var(setid)- is variable for setting setid that i get from database with my own scenario. IT does not matter. When running asterisk with weight 90 - all calls goes through it. When I starting asterisk with weight 10 -calls going through asterisk 90. When I shut down asterisk with weight 90 -calls goes through asterisk 10? but when i start asterisk weight 90 all calls goes through sterisk 10 until I shut down it. root@Kamailio:~# kamailio -v version: kamailio 4.3.0-dev3 (x86_64/linux) 8cdbe7 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 8cdbe7 compiled on 01:17:56 Jan 21 2015 with gcc 4.8.2 id setid destination flags priority attrs 1 2 sip:34.25.123.45:506000 0 weight=10 2 2 sip:10.0.1.6:506000 0weight=90 modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@10.0.1.1") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") modparam("tm", "reparse_on_dns_failover", 0) Thanks

2 3

NDB_REDIS password to REDIS remote DB
by Yuriy Gorlichenko 30 Jan '15

30 Jan '15

Hello. I try to use NDB_REDIS with remote REDIS DB and can not to connect because remote DB use password, but Kamailio module have no any variable or attr of modparam that implenets password for DB. How I can connect to my REDIS?

3 5

Jump to page:

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