sr-users November 2011

sr-users@lists.kamailio.org

78 participants
114 discussions

by Mokhtar Bengana

Hello, I am trying to install the tls module in 3.2.0. I run apt-get update before apt-get install kamailio-tls-modules and still can't load the tls module and getting the following error. Kamailio can't find the tls module even though it's installed? I didn't get any errors when I run apt-get install kamailio-tls-modules. Everything else is defined in the config file. Has someone tried to install tls in the latest release and is having issues? Any help much appreciated. #!define WITH_TLS #!ifdef WITH_TLS enable_tls=yes #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif root@Kamalio:/usr/local/lib/kamailio/modules_k# kamailio -c loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(12131) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <tls> in </usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/> 0(12131) : <core> [cfg.y:3501]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 268, column 12-19: failed to load module 0(12131) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <tls> found 0(12131) : <core> [cfg.y:3504]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 422, column 50: Can't set module parameter ERROR: bad config file (2 errors) root@Kamalio:/usr/local/lib/kamailio/modules_k# root@Kamalio:/usr/local/lib/kamailio/modules_k# apt-get install kamailio-tls-modules Reading package lists... Done Building dependency tree Reading state information... Done The following NEW packages will be installed: kamailio-tls-modules 0 upgraded, 1 newly installed, 0 to remove and 3 not upgraded. Need to get 0 B/399 kB of archives. After this operation, 848 kB of additional disk space will be used. Selecting previously deselected package kamailio-tls-modules. (Reading database ... 30034 files and directories currently installed.) Unpacking kamailio-tls-modules (from .../kamailio-tls-modules_3.2.0+squeeze1_i386.deb) ... Setting up kamailio-tls-modules (3.2.0+squeeze1) ... Thank you,

13 years, 3 months

Fwd: [Sip-implementors] FOSDEM 2012 Telephony/Communications Devroom Call for Presenters

by marius zbihlei

-------- Original Message -------- Subject: [Sip-implementors] FOSDEM 2012 Telephony/Communications Devroom Call for Presenters Date: Mon, 14 Nov 2011 20:28:53 +0100 From: Kevin P. Fleming <kpfleming(a)digium.com> To: sip-implementors(a)lists.cs.columbia.edu <sip-implementors(a)lists.cs.columbia.edu> Greetings, This is a call for presenters (with presentations!) for the telephony and communications devroom at FOSDEM 2012 - http://www.fosdem.org/. We will be holding a day full of presentations on development topics in the area of open source telephony and communications on Sunday, February 5th. The schedule allows for presentations from 9:00 to 17:00. The room we will be using will have a projector, wifi, and 100 seats. Please submit all proposals no later than 2011-12-09. Notification of accepted speakers will be provided by 2011-12-16. We will then work to have a schedule finalized by 2012-01-06. Talks should be submitted by subscribing to and then posting to the telephony-devroom mailing list hosted on http://lists.fosdem.org/. If you would like to contact the devroom organizer directly, please contact Kevin P. Fleming<kpfleming at digium.com>. Feel free to forward this along to any people or mailing lists that you think would be interested in this event. Thank you! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kfleming(a)digium.com | SIP: kpfleming(a)digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com& www.asterisk.org _______________________________________________ Sip-implementors mailing list Sip-implementors(a)lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors

13 years, 3 months

Re: [SR-Users] Compile kamailio modules only

by Daniel-Constantin Mierla

Hello, I think the problem was a comma at the end of sipcapture module name in kstandard group inside the Makefile. If you pull the branch 3.2, it should get the fix. The master was fixed by Ovidiu Sas few minutes ago when adding xhttp_rpc module in that list. Let us know if works now. Thanks, Daniel On 11/14/11 7:01 PM, Pavel Segeč wrote: > Hello, > > I'm trying to install homer capture agent on my Kamailio server. However for > the sipcapture module there is not precompiled package for debian 32 bit > neither 64 bit. Therefore I would like make module deb package and install > it after. > > I'm doing following steps: > > git clone --depth 1 git://git.sip-router.org/sip-router kamailio > cd kamailio/ > git checkout -b 3.2 origin/3.2 > make FLAVOUR=kamailio include_modules="sipcapture" cfg > cd pkg/kamailio/deb/ > mv debian debian-orig > ln -s lenny debian > cd ../../.. > make deb > > finaly after compilation I still have following packages only, > > kamailio > kamailio-postgres-modules_3.2.0_amd64.deb > kamailio_3.2.0_amd64.changes > kamailio-presence-modules_3.2.0_amd64.deb > kamailio_3.2.0_amd64.deb > kamailio-purple-modules_3.2.0_amd64.deb > kamailio_3.2.0.dsc > kamailio-python-modules_3.2.0_amd64.deb > kamailio_3.2.0.tar.gz > kamailio-radius-modules_3.2.0_amd64.deb > kamailio-berkeley-modules_3.2.0_amd64.deb > kamailio-snmpstats-modules_3.2.0_amd64.deb > kamailio-carrierroute-modules_3.2.0_amd64.deb > kamailio-sqlite-modules_3.2.0_amd64.deb > kamailio-cpl-modules_3.2.0_amd64.deb > kamailio-tls-modules_3.2.0_amd64.deb > kamailio-ldap-modules_3.2.0_amd64.deb > kamailio-unixodbc-modules_3.2.0_amd64.deb > kamailio-lua-modules_3.2.0_amd64.deb > kamailio-utils-modules_3.2.0_amd64.deb > kamailio-memcached-modules_3.2.0_amd64.deb > kamailio-xml-modules_3.2.0_amd64.deb > kamailio-mysql-modules_3.2.0_amd64.deb > kamailio-xmlrpc-modules_3.2.0_amd64.deb > kamailio-nth_3.2.0_amd64.deb > kamailio-xmpp-modules_3.2.0_amd64.deb > kamailio-perl-modules_3.2.0_amd64.deb > > without sipcapture module. I've tried to manually edit module.lst, manually > edit pkg/kamailio/deb/debian/rules, without results, still the same. What > I'm doing bad? > > thanks > > palo > > > > _______________________________________________ > 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://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

13 years, 3 months

TOS = 0

by marion deley

Hi, I'm trying to set tos value to zero but kamailio doesn't accept it. In fact, it's DSCP field which interest me but i can't set zero. Can you explain me why is it forbidden and how can i do that ? Thanks a lot Marion

13 years, 3 months

Problem with NAT and RTPProxy

by David Candamil Santos

Hi, I am using Kamailio 3.2.0 (x86_64/linux) and since some days ago I am trying to call from a network with NAT to outside. This is the configuration: Softphone (192.168.0.5) <--> Kamailio (192.168.0.3) <--> Router (192.168.0.1) <--> Softphone over smartphone All the router ports are opened and redirected to 192.168.0.3. I have installed rtpproxy 1.2.1-1 with the following configuration: --------------------------/etc/defaults/rtpproxy-------------------------------- # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 # Additional options that are passed to the daemon. EXTRA_OPTS="-l candamil.dyndns.org" ---------------------------------------------------------------------------------------- and works properly. This is the log message: ---------------------------------------------------------------------------------------------- Nov 12 20:09:13 condor kamailio[7001]: INFO: rtpproxy [rtpproxy.c:1415]: rtp proxy <udp:127.0.0.1:22222> found, support for it enabled ----------------------------------------------------------------------------------------------- Both softphones are "linphone". The configuration of the network softphone is the following: SIP identity: sip:1001@192.168.0.3 SIP proxy: sip:192.168.0.3 Direct connection to the Internet. In the softphone of the smartphone I set as proxy and domain " candamil.dyndns.org", the DNS address of my router IP. In this case, these are the symptoms: Both softphones can autentify correctly. When calling, both softphones ring, but when I answer, the voice signal is not received. This is the log error: ---------------------------------------------------------------------------------------------- Nov 12 20:23:14 condor kamailio[6991]: ERROR: rtpproxy [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy ---------------------------------------------------------------------------------------------- The same happens if I set in the network softphone that it's behind NAT, and I set the router IP. The same happens if I set in the network softphone that it's behind NAT and with a STUN server (stunserver.org). In the three cases, in the softphone over smartphone, the caller is 1001(a)192.168.0.3, and in the network softphone, the caller is 1002(a)candamil.dyndns.org. If I set in the network softphone the proxy as sip:candamil.dyndns.org, the same happens. If what I do is setting the SIP identity as sip:1001@candamil.dyndns.org, when calling form inside to ouside the network, the softphone doesn't know that the call was answered, and there is the following error in the log: ---------------------------------------------------------------------------------------------------------- Nov 12 20:53:00 condor kamailio[7306]: ERROR: <core> [parser/parse_via.c:2600]: ERROR: parse_via: invalid port number <5060ranch=z9hG4bKc 50f.b4825246.0> Nov 12 20:53:00 condor kamailio[7306]: ERROR: <core> [parser/parse_via.c:2629]: ERROR: parse_via on: <SIP/2.0/UDP 192.168.0.3:5060ranch=z 9hG4bKc50f.b4825246.0;received=87.223.138.84#015#012Via: SIP/2.0/UDP 87.223.138.84:5060;rport=5060;branch=z9hG4bK1021772993#015#012From: <sip:1001@candamil.dyndns.org>;tag=783852345#015#012To: < sip:1002@candamil.dyndns.org>#015#012Call-ID: 1644787160#015#012CSeq: 21 INVITE# 015#012User-Agent: Linphone/3.4.0 (eXosip2/unknown)#015#012Content-Length: 0#015#012#015#012> ------------------------------------------------------------------------------------------------------------ If the call is from outside to inside the network, it happens the same than in the previous cases. This is the relevant kamailio configuration: -----------------------------------------kamailio.cfg---------------------------------------------------- #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_NAT ####### 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. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!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 ######### /* 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="candamil.dyndns.org" /* 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 tcp_connection_lifetime=3605 ####### Custom Parameters ######### #!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" #!endif ####### Modules Section ######## #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- 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) #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); 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()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { 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 t_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(); exit; } } # USER location service route[LOCATION] { #!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 $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); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!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 if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control 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(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # 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; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$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); } # 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")) { route(TOVOICEMAIL); exit; } #!endif } -------------------------------------------------------------------------------------------------------------- I hope you can help me. Thanks for your time.

13 years, 3 months

Can the SBC (Kamailio 3.1.x and FreeSWITCH 1.0.6+) do Upper Registration?

by edson.gomes.leme

13 years, 3 months

SnmpStats problem

by Adam Kuśmirek

Hi All I've compiled kamailio 3.2.0 with snmpstats module. When trying to start the module I received the following message: 0(28197) DEBUG: <core> [sr_module.c:515]: load_module: trying to load </usr/lib/kamailio/modules_k/ snmpstats.so> 0(28197) ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </usr/lib/kamailio/modules_k/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefined symbol: boot_DynaLoader I run it on CentOS 5.7 After digging Internet, I found the solution for the problem. I had to modify Makefile in snmpstats module's directory and recompile it. I had to change BUILDAGENTLIBS env param before change: BUILDAGENTLIBS =-L$(LOCALBASE)/lib -lnetsnmpmibs -lnetsnmpagent \ -lnetsnmphelpers -lnetsnmp after change: BUILDAGENTLIBS =-L$(LOCALBASE)/lib -lnetsnmpmibs -lnetsnmpagent \ -lnetsnmphelpers -lnetsnmp \ -Wl,-E -Wl,-rpath,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE \ -L/usr/local/lib /usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a \ -L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE \ -lperl -lresolv -lnsl -ldl -lm -lcrypt -lutil -lpthread -lc \ -lwrap \ -lsensors \ -lrpmdb -lrpm After that I successfully started snmpstats module. I don't know why I didn't have compilation errors before adding this linking info, but got problem when trying to run compiled module. Can you explain it ? Do you know another way to solve the problem ? Regards Adam

13 years, 3 months

a2billing

by roman dmytriv

Hello folks, Any one tried using a2billing with Kamailio and Asterisk instead of Siremis? Can you share your experiens? Regards roman

13 years, 3 months

tracking ACC for a 486

by Sven Knoblich

Hi, I have a short question regarding a tm callback. I want to track a local generated ACK during a cfa. scenario: A calls B with a cfa to C B response with a 486 Busy this will be confirmed with an ACK. I saw, that TMCB_ACK_NEG_IN will only be called when a new transaction is created (t_lookup.c:t_newtran). Unfortunately in this case an ACK will be send directly when a reply will be received (t_reply.c:reply_received). Any ideas? thanks in advance, Sven

13 years, 3 months

segmentation fault in init_mod_child function

by Bruno Bresciani

Hi All, Kamailio 3.1.2 terminated with signal 11, segmentation fault, in the init_mod_child function at sr_module.c file. This problem occurs only when I try loading the loadbalance.so module, it was written by me. Someone Can help me to understand why this segmentation fault was generated? There are two other module written by me and this problem doesn't happen. Kamailio's trace: 9(27616) DEBUG: db_postgres [km_pg_con.c:80]: PQsetdbLogin(0x8db2878) 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): sl 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): auth_db 9(27616) DEBUG: <core> [db.c:294]: connection 0x82f4108 found in pool 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): usrloc 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): mi_fifo 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): registrar 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): nathelper 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): permissions 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): loadbalance 9(27616) DEBUG: loadbalance [loadbalance_mod.c:271]: loadbalance: child started! (5) 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): route 0(27582) ALERT: <core> [main.c:741]: child process 27585 exited by a signal 11 0(27582) ALERT: <core> [main.c:744]: core was generated 0(27582) INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD 8(27615) INFO: <core> [main.c:807]: INFO: signal 15 received 8(27615) DEBUG: <core> [main.c:818]: Memory status (pkg): 8(27615) DEBUG: fm_status: fm_status (0x82aefc0): 8(27615) DEBUG: fm_status: heap size= 4194304 8(27615) DEBUG: fm_status: used= 236720, used+overhead=264064, free=3930240 8(27615) DEBUG: fm_status: max used (+overhead)= 299824 8(27615) DEBUG: fm_status: dumping free list: 8(27615) DEBUG: fm_status: hash = 1 fragments no.: 27, unused: 0 bucket size: 8 - 8 (first 8) 8(27615) DEBUG: fm_status: hash = 44 fragments no.: 1, unused: 0 bucket size: 352 - 352 (first 352) 8(27615) DEBUG: fm_status: hash = 74 fragments no.: 1, unused: 0 bucket size: 592 - 592 (first 592) 8(27615) DEBUG: fm_status: hash = 77 fragments no.: 1, unused: 0 bucket size: 616 - 616 (first 616) 8(27615) DEBUG: fm_status: hash = 81 fragments no.: 1, unused: 0 bucket size: 648 - 648 (first 648) 8(27615) DEBUG: fm_status: hash = 84 fragments no.: 1, unused: 0 bucket size: 672 - 672 (first 672) 8(27615) DEBUG: fm_status: hash = 113 fragments no.: 51, unused: 0 bucket size: 904 - 904 (first 904) 8(27615) DEBUG: fm_status: hash = 2056 fragments no.: 1, unused: 0 gdb bt full trace: Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/route.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/route.so Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/rtpproxy.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/rtpproxy.so Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/tls.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/tls.so Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/perms_db.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/perms_db.so Reading symbols from /lib/libnss_files.so.2...done. Loaded symbols for /lib/libnss_files.so.2 Reading symbols from /lib/libnss_dns.so.2...done. Loaded symbols for /lib/libnss_dns.so.2 Core was generated by `/home2/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid'. Program terminated with signal 11, Segmentation fault. #0 init_mod_child (m=0xa238, rank=2) at sr_module.c:827 827 sr_module.c: Arquivo ou diretório não encontrado. in sr_module.c (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) bt full #0 init_mod_child (m=0xa238, rank=2) at sr_module.c:827 No locals. #1 0x00000000 in ?? () No symbol table info available. (gdb) Best Regards

13 years, 3 months

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sr-users November 2011