- sr-users - kamailio.org

How to configure Kamailio in stateful mode ?
by Luciana Oliveira 28 Nov '14

28 Nov '14

Please, help me. I tried several times and i did not find any way to configure kamailio to start it in stateful mode. Anyone know how to do it? Thanks, Luciana.

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xavp "contact_avp" question
by Julia Boudniatsky 28 Nov '14

28 Nov '14

Hello, We use get_redirects(*) and t_load_contacts in script. modparam("tm", "contacts_avp", "$avp(contacts)") In versions previous to 4.1 print all contacts: $(avp(contacts[*]) delete all contacts: $(avp(contacts[*])=$null >From 4.1 this avp is changed to xavp structure. What is the structure name and value name of xavp for print in log $xavp(contacts=>?) ? Is it possible to delete xavp? Thank you, Julia.

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New developer: Eloy Coto Pereiro
by Daniel-Constantin Mierla 28 Nov '14

28 Nov '14

Hello, I want to announce that a new person got developer GIT write access to repository: Eloy Coto Pereiro. He has developed a new module named statsd, which offers a connector from kamailio.cfg to manage values in statsd. The module will be merge to main repository, now is available at: - https://github.com/eloycoto/statsd His git commit id is: eloycoto My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Shifting to github for git repository
by Daniel-Constantin Mierla 28 Nov '14

28 Nov '14

Hello, (cross posting, looking to get feedback for those with more experience on github options - playing the lazy guy, not searching much on the web so far). In order to move the main git repository to github (where the commits will be pushed by devs) and have a real-time mirror on our server (stored very likely on kamailio.org, but we will keep git.sip-router.org as domain for it), two things have to be sorted out: 1) real time propagation of commits from github to git.sip-router.org I discovered that github provides a web hook for this event - when a commit is pushed, then a http query is done to a given url. I thought of using it to trigger the mirroring from git.sip-router.org I couldn't find if we can use what Victor Seva configured on git.sip-router.org to push in real-time to github, which I guess is a script invoked on commit -- if we can turn that around, would be probably the easiest. 2) email notifications Anyone with extensive knowledge of how much the notification messages can be customized? Eventually trying to get the structure as much as possible to what we have now -- see an example: - http://lists.sip-router.org/pipermail/sr-dev/2014-November/026253.html I could notice that Jitsi (using github) commit notification has a format not far from ours: - http://lists.jitsi.org/pipermail/commits/2014-November/012937.html I can eventually talk with them if no one else here has related knowledge. Another option is to trigger the notifications from the mirror repository we will keep on git.sip-router.org Also, would be good to know if there are people with some spare time willing to help whenever needed with this migration. Reply here to have you in mind. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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CNXCC PrePaid module
by Rene Montilva 27 Nov '14

27 Nov '14

Hi list I don't understand prepaid logic calculation, because when my credit is 0 or lees than cost per second, the call is established equal $var(i_pulse) and $var(f_pulse) how this works????

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WebRTC + Kamailio as SIP-proxy + rtpproxy(?) as media-proxy
by Rahul MathuR 27 Nov '14

27 Nov '14

Hello, I'm new to WebRTC although I've been using kamailio as sip proxy server for few months now. What I really do not know and trying to understand is - a) Can kamailio be used as sip-proxy while using WebRTC based UA calling to plain UAC/WebRTC based UAC ? b) What to use for media proxying (this really baffles me..) rtpproxy or rtpengine (?) or mediaproxy or rtpproxy-ng ? Is there any relation between them anywhere? c) I am not behind NAT and do not want secure web-sockets, so any sample config I can refer to ? d) Most likely, I'd be dealing with WebRTC <----->kamailio <-------> Freeswitch, but any pointers for WebRTC UAC to WebRTC based UAC or normal UAC would really be helpful. Kindly accept my thanks in advance for this !! -- Warm Regds. MathuRahul

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Some Questions about Kamailio UDP Transport and Concurrency
by Tony Peng 27 Nov '14

27 Nov '14

Dear Kamailio Developers： Hi. I have some questions about kamailio, and want to gain some suggestions from you. 1、The 4 UDP child processes are using system resources(CPU、memory), but they are using only one CPU core(I use 8 core CPU). Why? 2、The ablity of handling the Concurrency is alike, no matter how many UDP child processes I use. Why? And what's the effection of the other UDP child processes?

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Some Questions about Kamailio UDP Transport and Concurrency
by Tony Peng 27 Nov '14

27 Nov '14

Dear Kamailio Developers： Hi. I have some questions about kamailio, and want to gain some suggestions from you. 1、The 4 UDP child processes are using system resources(CPU、memory), but they are using only one CPU core(I use 8 core CPU). Why? 2、The ablity of handling the Concurrency is alike, no matter how many UDP child processes I use. Why? And what's the effection of the other UDP child processes?

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Dispatcher BYE issue
by Ahmed Salem 26 Nov '14

26 Nov '14

Hi, I'm trying to achieve this configuration phone --> NAT --> kamailio loadbalancer --> kamailio sip/proxy --> kamailio sip/proxy With the attached configuration I can make calls without any problems but when the callee hangs up the call does not end only the caller can hang up. Btw I'm using tcp for signalling. Any ideas? Thanks #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 mhomed=1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 tcp_async=yes tcp_connection_lifetime=3605 port=5060 sip_warning=yes ####### Modules Section ######## #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!endif loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "dispatcher.so" loadmodule "nathelper.so" loadmodule "rtpproxy.so" loadmodule "path.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 0) # ----- tm params ----- modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 3000) modparam("tm", "fr_inv_timer", 120000) # ----- dispatcher params ----- modparam("dispatcher", "db_url", "mysql://kamailio:kamailiorw@localhost/kamailio") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) #modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "flags_col", "flags") modparam("dispatcher", "priority_col", "priority") modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("path", "use_received", 1) modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) ####### Routing Logic ######## # main request routing logic request_route { xlog("L_NOTICE", "*** loose routing enforced before: $rm, $ru, $si, $du\n"); # per request initial checks route(REQINIT); loose_route(); # 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(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers #remove_hf("Route"); if (is_method("INVITE")){ record_route(); } if(is_method("REGISTER")){ add_path_received(); fix_nated_register(); } else{ add_contact_alias(); } # dispatch destinations route(DISPATCH); route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("L_NOTICE", "*** loose routing enforced before: $rm, $ru, $si, $du\n"); 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(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests #route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be 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; } } # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_domain("1", "0")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); return; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_domain()) { xlog("L_NOTICE", "going to next dst\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } }

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Configuring TLS and WSS with Kamailio
by Kamrul Khan 25 Nov '14

25 Nov '14

Hi, Im trying to connect my WebRTC clietn to kamailio via WSS. I successfully connected it via WS but having trouble connecting it via WSS. My first issue is I get error messages while i try to compile TLS module(console log in the end of the document). But, still it creates tls.so file. So i copied the tls.so to my kamailio modules directory and then updated my kamailio configuration as below: #!define WITH_TLS . .listen=tcp:127.0.0.1:5061 listen=tcp:192.168.146.133:5061 . . #!ifdef WITH_TLS enable_tls=yes #!endif . . #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/kamailio-devel/etc/kamailio/tls.cfg") #!endif Now, according to this mailing list,http://lists.sip-router.org/pipermail/sr-users/2013-March/077182.html : “when tls module is installed, a self signed pair of certificate-private key is generated in /usr/local/etc/kamailio” In my case, I dont even have the /usr/local/etc/kamailio directory. So, I followed http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates to create my certificate and key. And updated my tsl.cfg, this is how my tsl.cfg looks like: [server:192.168.146.133:5061] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/certs/sip.192.168.146.133/key.pem certificate = /etc/certs/sip.192.168.146.133/cert.pem ca_list = /etc/certs/demoCA/cert.pem [client:192.168.146.133:5061] verify_certificate = yes require_certificate = yes Then I restarted my kamailio server. And i get the following errors in my kamailio log: Im not sure what wrong im doing. Please help me: Oct 17 15:44:50 ubuntu kamailio: INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... Oct 17 15:44:50 ubuntu kamailio: WARNING: <core> [socket_info.c:1397]: fix_hostname(): WARNING: fix_hostname: could not rev. resolve 192.168.146.133 Oct 17 15:44:50 ubuntu kamailio: message repeated 2 times: [ WARNING: <core> [socket_info.c:1397]: fix_hostname(): WARNING: fix_hostname: could not rev. resolve 192.168.146.133] Oct 17 15:44:50 ubuntu kamailio: INFO: <core> [tcp_main.c:4836]: init_tcp(): init_tcp: using epoll_lt as the io watch method (auto detected) Oct 17 15:44:50 ubuntu kamailio: WARNING: <core> [daemonize.c:352]: daemonize(): pid file contains old pid, replacing pid Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: INFO: rr [rr_mod.c:159]: mod_init(): outbound module not available Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 512 Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: INFO: utils [utils.c:288]: mod_init(): forward functionality disabled Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: INFO: utils [utils.c:197]: pres_db_init(): xcap_auth_status function is disabled Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: ERROR: tls [tls_init.c:668]: tls_check_sockets(): TLSs<192.168.146.133:5061>: No listening socket found Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: ERROR: <core> [sr_module.c:970]: init_mod(): init_mod(): Error while initializing module tls (/usr/local/kamailio-devel/lib64/kamailio/modules/tls.so) Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: : tls [tls_locking.c:103]: locking_f(): BUG: tls: locking_f (callback): invalid lock number: 30 (range 0 - 0), called from eng_table.c:227 Oct 17 15:44:51 ubuntu kamailio: ERROR: <core> [daemonize.c:307]: daemonize(): Main process exited before writing to pipe tls module compile log xxx@ubuntu:/usr/local/src/kamailio-4.1/kamailio$ sudo make -C modules/tls make: Entering directory `/usr/local/src/kamailio-4.1/kamailio/modules/tls' make: Leaving directory `/usr/local/src/kamailio-4.1/kamailio/modules/tls' make: Entering directory `/usr/local/src/kamailio-4.1/kamailio/modules/tls' CC (gcc) [M tls.so] tls_bio.o CC (gcc) [M tls.so] tls_cfg.o CC (gcc) [M tls.so] tls_config.o CC (gcc) [M tls.so] tls_ct_wrq.o CC (gcc) [M tls.so] tls_domain.o In file included from tls_domain.c:39:0: tls_domain.c: In function âload_certâ: tls_util.h:52:6: warning: variable âretâ set but not used [-Wunused-but-set-variable] int ret; \ ^ tls_domain.c:506:4: note: in expansion of macro âTLS_ERRâ TLS_ERR("load_cert:"); ^ tls_domain.c: In function âload_ca_listâ: tls_util.h:52:6: warning: variable âretâ set but not used [-Wunused-but-set-variable] int ret; \ ^ tls_domain.c:536:4: note: in expansion of macro âTLS_ERRâ TLS_ERR("load_ca_list:"); ^ tls_util.h:52:6: warning: variable âretâ set but not used [-Wunused-but-set-variable] int ret; \ ^ tls_domain.c:543:4: note: in expansion of macro âTLS_ERRâ TLS_ERR("load_ca_list:"); ^ tls_domain.c: In function âload_crlâ: tls_util.h:52:6: warning: variable âretâ set but not used [-Wunused-but-set-variable] int ret; \ ^ tls_domain.c:575:4: note: in expansion of macro âTLS_ERRâ TLS_ERR("load_crl:"); ^ tls_domain.c: In function âload_private_keyâ: tls_util.h:52:6: warning: variable âretâ set but not used [-Wunused-but-set-variable] int ret; \ ^ tls_domain.c:990:5: note: in expansion of macro âTLS_ERRâ TLS_ERR("load_private_key:"); ^ tls_util.h:52:6: warning: variable âretâ set but not used [-Wunused-but-set-variable] int ret; \ ^ tls_domain.c:998:4: note: in expansion of macro âTLS_ERRâ TLS_ERR("load_private_key:"); ^ tls_util.h:52:6: warning: variable âretâ set but not used [-Wunused-but-set-variable] int ret; \ ^ tls_domain.c:1005:4: note: in expansion of macro âTLS_ERRâ TLS_ERR("load_private_key:"); ^ CC (gcc) [M tls.so] tls_dump_vf.o CC (gcc) [M tls.so] tls_init.o CC (gcc) [M tls.so] tls_locking.o CC (gcc) [M tls.so] tls_mod.o CC (gcc) [M tls.so] tls_rpc.o CC (gcc) [M tls.so] tls_select.o CC (gcc) [M tls.so] tls_server.o In file included from tls_server.c:52:0: tls_server.c: In function âtls_complete_initâ: tls_util.h:52:6: warning: variable âretâ set but not used [-Wunused-but-set-variable] int ret; \ ^ tls_server.c:192:3: note: in expansion of macro âTLS_ERRâ TLS_ERR("Failed to create SSL or BIO structure:"); ^ tls_server.c: In function âtls_shutdownâ: tls_util.h:52:6: warning: variable âretâ set but not used [-Wunused-but-set-variable] int ret; \ ^ tls_server.c:521:4: note: in expansion of macro âTLS_ERRâ TLS_ERR("SSL error:"); ^ tls_server.c: In function âtls_encode_fâ: tls_util.h:52:6: warning: variable âretâ set but not used [-Wunused-but-set-variable] int ret; \ ^ tls_server.c:798:5: note: in expansion of macro âTLS_ERRâ TLS_ERR(err_src); ^ tls_util.h:52:6: warning: variable âretâ set but not used [-Wunused-but-set-variable] int ret; \ ^ tls_server.c:832:5: note: in expansion of macro âTLS_ERRâ TLS_ERR(err_src); ^ tls_server.c: In function âtls_read_fâ: tls_util.h:52:6: warning: variable âretâ set but not used [-Wunused-but-set-variable] int ret; \ ^ tls_server.c:1186:4: note: in expansion of macro âTLS_ERRâ TLS_ERR(err_src); ^ tls_util.h:52:6: warning: variable âretâ set but not used [-Wunused-but-set-variable] int ret; \ ^ tls_server.c:1220:4: note: in expansion of macro âTLS_ERRâ TLS_ERR(err_src); ^ CC (gcc) [M tls.so] tls_util.o CC (gcc) [M tls.so] tls_verify.o LD (gcc) [M tls.so] tls.so make: Leaving directory `/usr/local/src/kamailio-4.1/kamailio/modules/tls'

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Problems installation Kamailio Ubuntu 12.04
by kai.ohnacker＠cbc.de 25 Nov '14

25 Nov '14

Hello everybody, I tried to install Kamailio 4.1 with help of this instruction -> http://www.kamailio.org/wiki/install/4.1.x/git. But, if I type the command (Root:/usr/local/src/kamailio-4.1#) git clone --depth 1 git://git.sip-router.org/kamailio kamailio I got this message back fatal: unable to connect to git.sip-router.org: git.sip-router.org[0: 78.46.84.71]: errno=Connection timed out If I open the website git.sip-router.org with my webbrowser, I got a connection. Does somebody know what the problem might be? Best regards, Kai O.

3 2

General debugging and presence question
by Jesse S. Anderson 25 Nov '14

25 Nov '14

Hi All, Sorry for the general questions here, I'll try to be concise. I'm kind of a n00b, so bear with me. Problem: Kamailio will not respond or accept SUBSCRIBE. It kicks back a 404 every time. SUBSCRIBE is ok. Question #1: I actually copied a config from the vendor we are using and it's supposed to work so is there something simple I'm missing? Question #2: Logging. I've read that there should be a kamailio.log file somewhere that I can tail -f, but there's no such file on the machine. I've done a *kamailio debug -dd *but it's really tough to go through. Doesn't seem like you can grep using that either. If anyone out there has any advice for me, I would certainly appreciate it. Thanks!

3 2

sqlite load module error
by Javier Ricke 24 Nov '14

24 Nov '14

i have install this lirebries: * SQlite 3 library - e.g., libsqlite3-0 * SQlite 3 devel library - to compile the module e.g. libsqlite3-dev * SQlite 3 utilities - for kamdbctl e.g. sqlite3 but in /usr/local/lib/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/ dont have module db_sqlite.so any? regards!

2 2

host undefined
by Javier Ricke 24 Nov '14

24 Nov '14

------phona A-------kamailio---------asterisk----- serverjavier-desktop*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime JavierTren/1000 152.74.xxx.xxx D Yes Yes 5060 Unmonitored Cached RT 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline] serverjavier-desktop*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] in host appear sofphone's ip, but spend time and host is undefinid, therefore asterisk dont could route call... plz help :D

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Kamailio does not delete expired contacts
by Bui Minh Yen 24 Nov '14

24 Nov '14

Hi everyone, I am using Kamailio-4.1.5. I met a problem that Kamailio does not delete expired contact. I enable log file for module usrloc and see that, about 1 first hour, timer callback of usrloc module working fine (delete all expired contacts). But after that, it is not called --> Expired contacts in shared memory and database are not deleted. I configured usrloc module like that: # ----- usrloc params ----- modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "db_ops_ruid", 1) Could you help me to fix this problem! Thanks so much!

2 1

SNMP stats problem
by Muszalski Michał - Hurt 24 Nov '14

24 Nov '14

Good evening, I'm quite new in the Kamailio community so at the beginning I just want to say "Hello" to everyone. I have a following problem with SNMP in Kamailio. I want to collect data from Kamailio by SNMP. I've installed snmpd on the machine (Linux RHEL) and it's working. I've load snmpstats module in Kamailio and it starts up without error. I've configured Kamailio to use agentXsocket and enable agentx in SNMP. Port 705 is opened and netstat shows connection established between Kamailio and SNMP. Wireshark shows some packets going through. The problem is that there is no data during snmpwalk from Kamailio. Just standard data from SNMP, nothing from Kamailio. I've copied Kamailio MIBs to the proper directory, added it in the snmp.conf. Moreover I've tried to add -m +KAMAILIO-REG-MIB to snmpwalk. Still no data from Kamailio during snmpwalk. Any ideas? Best Regards, Michal Muszalski

3 2

RTP Between end points behind NAT or edit headers's sdp?
by Javier Ricke 24 Nov '14

24 Nov '14

------phona A-------kamailio---------asterisk----- OPTION 1: configure asterisk or kamailio i used asterisk, and install kamailio for traffic RTP can be send between end points that behind NAT router, and do not have to go through RTP proxy,, plz help! i think to the moment install kamailio, headrs'sdp fix IP private, but no!, how can fix it plz!!? regards! or OPTION 2: edit sip/sdp mi sip/sdp is [code] <--- SIP read from UDP:152.74.21.12:5060 ---> ACK sip:1001@152.74.21.12:6112 SIP/2.0 Via: SIP/2.0/UDP 152.74.21.12;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 190.164.204.227:41553 ;branch=z9hG4bK-d8754z-84f73c7e042445de-1---d8754z-;rport=41553 Max-Forwards: 16 Contact: <sip:JavierTren@190.164.204.227:41553;transport=UDP> To: <sip:1001@152.74.21.12;transport=UDP>;tag=as6e487bf1 From: <sip:JavierTren@152.74.21.12;transport=UDP>;tag=2b8fa52c Call-ID: NjgyMTViMDAyMzczNjIyNWIwZWU3OWJjZDAxMWFkNjY. CSeq: 2 ACK Proxy-Authorization: Digest username="JavierTren",realm="152.74.21.12",nonce="VGFnblRhZkIpJRpScSaEi795VKe4uof0",uri=" sip:1001@152.74.21.12 ;transport=UDP",response="116bf459c22231d0a770534d674b768d",algorithm=MD5 User-Agent: Zoiper r27147 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- set_destination: Parsing <sip:152.74.21.12;lr=on;ftag=2b8fa52c> for address/port to send to set_destination: set destination to 152.74.21.12:5060 Audio is at 12064 Adding codec 100002 (gsm) to SDP Adding codec 100003 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 152.74.21.12:5060: INVITE sip:JavierTren@190.164.204.227:41553;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 152.74.21.12:6112;branch=z9hG4bK3992226d;rport Route: <sip:152.74.21.12;lr=on;ftag=2b8fa52c> Max-Forwards: 70 From: <sip:1001@152.74.21.12;transport=UDP>;tag=as6e487bf1 To: <sip:JavierTren@152.74.21.12;transport=UDP>;tag=2b8fa52c Contact: <sip:1001@152.74.21.12:6112> Call-ID: NjgyMTViMDAyMzczNjIyNWIwZWU3OWJjZDAxMWFkNjY. CSeq: 102 INVITE User-Agent: Asterisk PBX 11.13.0 Session-Expires: 1800;refresher=uac Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 256 v=0 o=root 1842142539 1842142540 IN IP4 192.168.1.8 s=Asterisk PBX 11.13.0 c=IN IP4 192.168.1.8 t=0 0 m=audio 8000 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 [/code] and the field o and c i need to IP public y no private,, plz any?

2 1

database scheme generation and synchronization
by Måns Nilsson 21 Nov '14

21 Nov '14

Greetings, I'm fighting a number of errors in the 'acc' table like so: Nov 21 01:43:05 landsort postgres[53327]: [4-1] ERROR: column "time" of relation "acc" does not exist at character 78 That is, I believe, the default column name. Thing is, though, that the DDL that I used to build the table, is the one in in scripts/ and it does not does not contain that string: (on master but it is the same in 4.2 which is what I've got in production) $ for d in mysql oracle postgres ; do egrep '\btime\b' scripts/${d}/*create* ; done $ How are those files generated? How do I come about to submitting an useful patch? What is the desired datatype for the time column? -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 Edwin Meese made me wear CORDOVANS!!

2 2

rtpengine flags UDP/TLS/RTP/SAVP and UDP/TLS/RTP/SAVPF
by Juha Heinanen 21 Nov '14

21 Nov '14

Richard Fuchs wrote: > The command flags "RTP/AVP" etc are translated into the key > "transport-protocol=$X" within the control protocol. If you wish to > force usage of one of the UDP/TLS/... protocol, you should be able to do > so by spelling it out in your command. I believe this is not documented, > but should work. i tried and the offer and audio worked accordingly: Nov 21 05:37:15 rautu /usr/bin/sip-proxy[11113]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1 transport-protocol=UDP/TLS/RTP/SAVP trust-address) for consistency, it might be a good idea to make the dtls flags known to rtpengine module. -- juha

1 0

Kamailio creates tons of REDIRECT packages...
by Arnau Riba 20 Nov '14

20 Nov '14

Hello! I'm having this issue and I'm not able to solve it. I have searched it many times on the Internet but I found nothing, that's why I am asking here... I'm sorry I do not wanna cause any troubles but I need to get this done... This is my scenario, I want to make a call through a Kamailio server that redirects to another server. I have an User A registered to Server A and a User B registered to Server B, what I want to do is the following: User A calls User B using Server A, because Server A doesn't have the User B the Server A will redirect to Server B and make the call. I'm using Ekiga and of course Kamailio (and I want to make it work both ways). This is the code I'm using to do this in kamailio.cfg, I replaced this lines of code: *$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; } }* For this ones: *$avp(oexten) = $rU; if (!lookup("location")) { sl_send_reply("300",("REDIRECT")); rewritehostport("10.80.129.177"); forward(uri:host,uri:port); break; }* And it didn't work, then I putted the lines above as the following: *$avp(oexten) = $rU; if (!lookup("location")) { * * forward(uri:host,uri:port); rewritehostport("10.80.129.177");* * sl_send_reply("300",("REDIRECT")); break; }* And it worked but if I do a Wireshark Capture on the Server I get a lot of REDIRECT packets and my teacher says this is wrong, that I should only get 1. He did not help me and said I must learn and I want but I'm not able to understand what's going on, I need to do this because I want to pass the exam and not to fail... Again I'm sorry for taking your time, I'm asshamed. Thank you very much

2 1

Re: [SR-Users] rtpengine UDP/TLS/RTP/SAVP <=> RTP/AVP re-INVITE
by Richard Fuchs 20 Nov '14

20 Nov '14

On 11/18/2014 08:29 PM, Juha Heinanen wrote: > Richard, > > Enclosed find syslog that includes rtgpengine log at level 7 and also > pcap of the call. I made re-invite from baresip to sems once I saw this > in syslog: > > Nov 19 03:22:14 rautu rtpengine[29718]: [5315d797a9a5e7ce port 50761] DTLS-SRTP successfully negotiated > > I didn't include the to this message due to the pcap. Looking at your logs, I think what happened is that you've included the "trust-address" flag in the original answer, but haven't included it in the answer to the re-invite. This means that in the second answer, rtpengine erroneously saw a different endpoint than before (using the source address of the SIP packet), in response to this it also opened a new endpoint on its own side, which was then sent to the DTLS-SRTP client. This client then saw the new endpoint and correctly started a new DTLS connection, meanwhile rtpengine was still using the old keys and (I'm guessing) neither expected nor processed the new DTLS handshake, thus en/decrypted SRTP using the old keys, resulting in mangled RTP packets. So the short-term fix would be to include trust-address to avoid switching endpoints. As for DTLS restarts in rtpengine, this is something that I was just working on and DTLS restarts on changed endpoints should be working soon. cheers

2 1

Thread safety of shared variables
by Alex Balashov 19 Nov '14

19 Nov '14

Does setting $shv()s in script require lock()ing, or is it inherently thread-safe? Thanks! -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

3 5

Kamailio Does NOT Forward Registration Requests To Asterisk.
by Mahmoud Ramadan Ali 19 Nov '14

19 Nov '14

Hi Dears, I'm trying to configure Kamailio as SBC in multi home mode for Asterisk by authenticating the inbound SIP registration requests,i'm following this tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb to achieve this goal. i have modified the necessary changes like the Asterisk DB URL and the SIP table name and Username and password column and verified the connection. My topology like this *Asterisk (192.168.100.10) <----Internal:192.168.100.1---->Kamailio<---External:192.168.50.1-----> SIP Phone (192.168.50.2)* But when trying to register a SIP phone Kamailio does NOT forward the authentication request to Asterisk and sends 401 Unauthorized error message.I've attached my config file if any one wants to check it and thanks in advance. Best Regards U 192.168.50.2:37297 -> 192.168.50.1:5060 REGISTER sip:192.168.50.1;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 192.168.50.2:37297 ;branch=z9hG4bK-d8754z-a46e0c7c9d98fe52-1---d8754z-;rport;transport=UDP. Max-Forwards: 70. Contact: <sip:1001@192.168.50.2:37297 ;rinstance=1d7c44dbcb8a7a2f;transport=UDP>. To: <sip:1001@192.168.50.1;transport=UDP>. From: <sip:1001@192.168.50.1;transport=UDP>;tag=1d222e19. Call-ID: NTc2NDBjMGQ2YWFmZjdmNWI0MzVmN2Y4NzYyODJlMTc.. CSeq: 2 REGISTER. Expires: 70. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE. Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri. User-Agent: Z 3.2.21357 r21367. Authorization: Digest username="1001",realm="192.168.50.1",nonce="VGfAuFRnv4wMvoTG7wA9tqYD9fgZDe3D",uri="sip:192.168.50.1;transport=UDP",response="8bbd01d879250585eafee4f510689f73",algorithm=MD5. Allow-Events: presence, kpml. Content-Length: 0. # U 192.168.50.1:5060 -> 192.168.50.2:37297 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.50.2:37297 ;branch=z9hG4bK-d8754z-a46e0c7c9d98fe52-1---d8754z-;rport=37297;transport=UDP. To: <sip:1001@192.168.50.1 ;transport=UDP>;tag=b27e1a1d33761e85846fc98f5f3a7e58.fe8b. From: <sip:1001@192.168.50.1;transport=UDP>;tag=1d222e19. Call-ID: NTc2NDBjMGQ2YWFmZjdmNWI0MzVmN2Y4NzYyODJlMTc.. CSeq: 2 REGISTER. WWW-Authenticate: Digest realm="192.168.50.1", nonce="VGfAuFRnv4wMvoTG7wA9tqYD9fgZDe3D". Server: kamailio (4.1.6 (i386/linux)). Content-Length: 0.

2 6

rtpengine and sdes
by Daniel-Constantin Mierla 19 Nov '14

19 Nov '14

Hello, I saw that rtpengine docs still advertise support for SDES SRTP and I was wondering if anyone was (is still) using it for decryption/encryption of this type of SRTP, mainly in scenarios like SRTP from a classic sip phone (e.g., snom) to another SIP endpoint wihtout srtp support. As I cc-ed Richard, another question is about the plans for the future, if at this moment is properly working/maintained, will it be kept for the future or the plan is to deprecate/remove/don't invest any effort in its maintenance? Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Training/Tutorials?
by Jordan Rinke 19 Nov '14

19 Nov '14

For the life of me I can't find any in depth online resources on setting up and configuring Kamailio. Do anyone know of some good resources? Specifically I want to put this in front of a couple Asterisk servers so I need to learn how to connect a trunk to it, and then based on the call destination route it to the right Asterisk servers. I feel like this shouldn't be complicated I just don't understand where/how Kamailio is configured coming from only using Asterisk previously.

2 1

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