sr-users September 2014

sr-users@lists.kamailio.org

96 participants
170 discussions

by Tom Johnson

Is there a way to call the send_reply method of the route from either Java or csharp? As I have been having issues with the app_java, I have been looking at app_mono and it seems to have better access to the variables, etc. However, the SR.core.ModF only seems to take methods that do not take parameters. Or for that matter, is there a way to return something back to the route in the config file which it can use to call send_reply? I have a strange requirement to return 202 rather than 200 from subscription requests and a 603 if it fails. Tom Johnson<mailto:tjohnson@microautomation.com> | Senior Software Engineer O: 703-543-2118 | F: 703-543-2099 5870 Trinity Parkway, Suite 600, Centreville, VA 20120 [cid:image001.png@01CE18AF.8F231EB0]<http://www.microautomation.com/>

10 years, 2 months

app_java issue

by Tom Johnson

Per my previous message, I determined that the module's makefile was not properly finding the jdk/jre directories needed to compile. So, I temporarily hardcoded the paths in the makefile. However, now when I use the module, Kamailio crashes. Below is the lines from /var/log/messages: Sep 9 15:13:10 MAIFREEPBX kamailio: INFO: <core> [tcp_main.c:4836]: init_tcp(): init_tcp: using epoll_lt as the io watch method (auto detected) Sep 9 15:13:10 MAIFREEPBX kamailio: WARNING: <core> [daemonize.c:352]: daemonize(): pid file contains old pid, replacing pid Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: rr [rr_mod.c:159]: mod_init(): outbound module not available Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 512 Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: app_java [java_mod.c:125]: mod_init(): Initializing Java VM with options: -Xdebug -verbose:gc,class,jni -Djava.compiler=NONE -Djava.class.path=/opt/microautomation/i3/I3Notifier.jar:/opt/kamailio/java/kamailio.jar:/opt/microautomation/i3/lib/log4j-1.2.16.jar Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: app_java [java_mod.c:146]: mod_init(): app_java: Java VM initialization OK Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: app_java [java_mod.c:198]: mod_init(): app_java: module initialization OK Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: : <core> [mem/q_malloc.c:159]: qm_debug_frag(): BUG: qm_*: prev. fragm. tail overwritten(0, abcdefed)[0x7f0941207190:0x7f09412071c0]! Sep 9 15:13:11 MAIFREEPBX abrt[12282]: Saved core dump of pid 12268 (/usr/local/sbin/kamailio) to /var/spool/abrt/ccpp-2014-09-09-15:13:10-12268 (183721984 bytes) The code dump mentioned does not exist. If I do not load the app_java module, Kamailio works. Tom Johnson<mailto:tjohnson@microautomation.com> | Senior Software Engineer O: 703-543-2118 | F: 703-543-2099 5870 Trinity Parkway, Suite 600, Centreville, VA 20120 [cid:image001.png@01CE18AF.8F231EB0]<http://www.microautomation.com/>

10 years, 2 months

websocket:ws_current_connections vs. kamctl fifo ws.dump

by Luis F Urrea

Hello I have a question in regards monitoring websocket connections on a production box. We ususally trust whatever is indicated through the $ kamctl fifo get statistics We've seen that this is what the community contributed munin plugins use. However when trying to monitor web socket connections and using this output we've seen completely different results from the output of $ kamctl fifo get_statistics websocket:ws_current_connections Which seem to indicate current websocket connections and the output of $ kamctl fifo ws.dump Which seems to indicate current OPEN connections. Can anybody shed some light as to what are the differences between both if any, and advice which one is the proper one to monitor for possible file handle exhaustion. TIA

10 years, 2 months

Auto-response

by Mick Stevens

10 years, 2 months

Re: [SR-Users] TLS Handshake failing with WSS

by Manuel Camarg

According to this article I mentioned in the beginning: http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket In http:request they do straight away the set_reply_close(); I don't know exactly if this kamailio.cfg implementation may be useful for working with WSS: 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; } } HTTP authentication is not being properly set up. Any indications on how to perform this sucessfully? Kind regards Manuel >Hello, > >from the logs you sent now, it appears that you have set_reply_close() >in config, therefore the connection is closed after sending the reply. > >Cheers, >Daniel

10 years, 2 months

How to increment dialog calee_cseq value when using t_uac_dlg via xmlrpc for generating INFO requests

by Drazen Rep

Hi, I am using Kamailio 4.1.1 and dlg_manage() function for processing SIP requests with dialog module. After call is established Kamailio is being used for generating in-dialog INFO request toward caller (using t_uac_dlg function via xmlrpc). INFO messages are correctly generated but dialog calee_cseq value is not being incremented with every message. That represents a problem since other SIP requests generated by callee side (like OPTIONS) are not in order, and there is no correct information about calee_cseq value in dialog memory records. The question is how can I increment dialog calee_cseq value when local INFO request toward caller is being generated using t_uac_dlg via xmlrpc? Thx, Drazen

10 years, 2 months

UAC module

by Yuriy Gorlichenko

Hello. I tried to use UAC module at my system for registration multiple providers, and call through it. When I implementing this at my cfg file I had some troubles: -- with Authentification at INVITE (using UAC_AUTH). IT does not send second INVITE packet with www_auth header for some trunks (not for all, only for some providers) -- with BYE recieving from PROVIDER ("No BYE from Called party" task at USers Mailing lists) After many days of tests and debugging I ask question from myself - does module correctly works? As I know this module breaks some philosophy about SIP proxy and development of this module is not question of first priority. After that all I need advice - what I must use for registtration my SIP trunks - 3d party registrant or this module, that works fine and I doing something wrong? Thanks for advice.

10 years, 2 months

Kamailio v4.2 - development frozen

by Daniel-Constantin Mierla

Hello, development (master) branch of Kamailio enters now in pre-release phase for version 4.2.0. Therefore, no new feature should be pushed to master until we create a dedicated branch for 4.2 (expected to be in about 4 weeks or so). If in doubt to push or not a commit to master, push it first on a personal branch (or attach to an email) and discuss it on sr-dev. The new modules can be a bit more dynamic if there is need to get them to the right shape (e.g., like decision to rename functions, parameters or adjust database structure). We hope to get many people involved in testing, to rich a stable state before releasing 4.2.0. If you want to get involved and need assistance, don't hesitate to write to mailing lists. Help with updating the wiki page for migration from 4.1 to 4.2 as well as what is new in 4.2 is very appreciated. I will post updates about the very soon. Many thanks to everyone involved in development of 4.2 and the early testers that played with master branch. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

10 years, 2 months

LCR base on call rate

by Satish Patel

I heard somewhere LCR can do routing based on call rate and call price, does it true? I haven't seen any config or doc which does call rate using LCR. It only does routing base on prefix scan. Am I missing something here? Sent from my iPhone

10 years, 2 months

WebRTC meeting Norfolk, 15 October 2014

by Daniel Pocock

I'll be in Norfolk, VA for xTupleCon in October On 15 October, there will be two events for WebRTC: 14:15 a talk about the xTuple WebRTC extension at xTupleCon - must register for xTupleCon to attend this 17:30 a technical / developer workshop at xTuple's offices - free, anybody welcome, even if not attending xTupleCon, RSVP through Eventbrite[1] Please see my blog[2] for more comments about all of this and feel free to email me in advance if you have questions about it or if you may like to meet up there. 1. http://www.eventbrite.com/e/browser-based-webrtc-telephony-for-web-apps-wor… 2. http://danielpocock.com/xtuplecon-webrtc-talk-schedule-change

10 years, 2 months

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sr-users September 2014