sr-users November 2011

sr-users@lists.kamailio.org

78 participants
114 discussions

by Daniel-Constantin Mierla

Hello, I put together a wiki page to help people porting patches between branches, exemplified with master and 3.2 branches -- the typical patch backporting these days: * http://www.kamailio.org/wiki/devel/backporting-to-3.2.x They are useful also for people with no write access, in case they want to try patches committed to devel branch in their local git repo clones. In additions, I added to kamailio wiki the page with guidelines for commits, trying to give the generic suggestions of how to do it in order to keep the coherence of the development: * http://www.kamailio.org/wiki/devel/git-commit-guidelines Updates and enhancements are welcome! If one has more suggestions or want to debate some of the proposed guidelines, feel free to start the discussion on the mailing list. Cheers, Daniel -- 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

Limiting simultaneous calls

by Mino Haluz

Hi, I was using cdrtool (prepaid table) and callcontrol to limit concurrent calls. In fact this is only limiting the outbound calls, but I would like to use another mechanism which should limit the inbound calls too. So basically to limit voice channels. So is there some reliable method/module how to achieve this? Thanks, Mino

13 years

Dropped registrar bindings with 1.5.3

by Alex Balashov

I have a 1.5.3 installation functioning as a registrar that is exhibiting a very curious, if only infrequent set of behaviours. The host is CentOS 5.x, with PostgreSQL 9.0 backing usrloc, and db_mode = 1 (immediate write-through). I have a registration handler in my initial request route that begins with a log message: ... else if(is_method("REGISTER")) { xlog("L_INFO", "... Processing REGISTER from $si:$sp for AOR $tu\n"); route(2); exit; } I've got a registrant who consistently re-registers with an expiration time of 120, so in theory they ought to be re-registering every two minutes or so. Most of the time they do. But, occasionally, I'll see a gap of 5-8 minutes go by, in which the above log message does not appear, and their registration expires so that they can't receive calls. That's not what's interesting. What's interesting is that I did a packet capture on the proxy and caught one of these scenarios. Within that ~8 minute window when the registration expired and calls were failing, the UAC actually re-registered several times, on normal ~2 minute boundaries, AND the proxy successfully challenged the request and sent a 200 OK indicating that the binding was stored! I double-checked all the parameters to ensure that the 401s and 200 OKs corresponded to the right REGISTER flow, etc. - yes, it's all correct. Is there any imaginable set of circumstances or a known bug that would cause Kamailio to successfully authenticate and affirm a registration request, while neither logging its receipt, nor, evidently, storing it to the database? I can envision a number of database failure modes that would account for the binding not actually being in the 'location' table, but that doesn't explain why the request wouldn't even be logged. That's what has me baffled. Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

13 years

[Kamailio-Users] Converting rtpproxy recorded .rtp file to .wav file

by Vikram Ragukumar

Hello, I used Kamailio+rtpproxy to record a session and rtpproxy outputs the following files long_file_name.a.rtp, long_file_name.a.rtcp, long_file_name.o.rtp, long_file_name.o.rtcp http://www.rtpproxy.org/wiki/RTPproxy/FAQ From the Rtpproxy FAQ above, i tried to extract the audio using rtpbreak and sox. rtpbreak -W -r long_file_name.a.rtp rtpbreak -W -r long_file_name.o.rtp The above commands generate rtp.0.0.raw, rtp.1.0.raw. Then when i run sox using sox --combine merge -r 8k -A rtp.0.0.raw -r 8k -A rtp.1.0.raw -t wavpcm -s out.wav i get the following errors : sox: invalid option -- - sox: -c must be given a number Is there a switch/anything else that i am missing ? Thanks in advance, Regards, Vikram.

13 years

sqlops crash when connection does not exist

by Juha Heinanen

i misspelled connection argument in sql_query and got crash: Nov 10 17:19:15 sip /usr/sbin/sip-proxy[4182]: ERROR: sqlops [sqlops.c:266]: invalid connection [sip-proxy] Nov 10 17:19:15 sip /usr/sbin/sip-proxy[4182]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/sip-proxy/sip-proxy.cfg:281 Nov 10 17:19:15 sip kernel: [ 990.129668] sip-proxy[4182]: segfault at 786e6570 ip b769de38 sp bf960ffc error 4 in libc-2.11.2.so[b762b000+140000] (gdb) where #0 0xb769de38 in strcmp () from /lib/i686/cmov/libc.so.6 #1 0x080f54f9 in find_mod_export_record (mod=0xb750bb80 "db_mysql", name=0xb720d353 "db_bind_api", param_no=0, flags=0, mod_if_ver=0xbf96108c) at sr_module.c:657 #2 0x080f5756 in find_mod_export (mod=0xb750bb80 "db_mysql", name=0xb720d353 "db_bind_api", param_no=0, flags=0) at sr_module.c:722 #3 0xb720449c in db_bind_mod (mod=0xb6628944, mydbf=0x8) at db.c:209 #4 0xb66240d2 in ht_db_init_con () at ht_db.c:75 #5 0xb661b144 in destroy () at htable.c:225 #6 0x080f3684 in destroy_modules () at sr_module.c:782 #7 0x0809087c in cleanup (show_status=0) at main.c:564 #8 0x08091509 in shutdown_children (show_status=0, sig=<value optimized out>) at main.c:706 #9 0x080940f7 in main (argc=17, argv=0xbf961354) at main.c:2523 -- juha

13 years

Core compiled with Sctp support

by Stefan Ljung

I've downloaded sip-router 3.1.2 and have been struggling with the make commands for configuring. Compiling and installing. Can't really say that I master it. But I have finally managed to do all steps and started a example ser.cfg. One of the features I want to play around eith is the sctp transport. >From previous experiments with OpenSER I kearned that if I only specify a listen host address without transport protocol all three UDP TCP and SCTP sockets will be started. Ths time I only see UDP and TCP: Listening on udp: 127.0.0.1:5060 tcp: 127.0.0.1:5060 Aliases: tcp: localhost:5060 tcp: localhost.localdomain:5060 udp: localhost:5060 udp: localhost.localdomain:5060 I have read the c sources to verify that the sctp libs and header files are in the right places, and they are. What I could need is a tip on how to verify that my core has been compiled with sctp support. How can I see that ? In what files would there be a confimation of that the sctp feature will be compiled in? /Stefan

13 years

Problem trying to play recorded message before call

by David Candamil Santos

Hi there, guys. I hope you can help me. I use kamailio to comunicate some servers (each one with its own kamailio). When an user calls another, depending on the userID that will receive the call, the server redirects the call to the corresponding kamailio server. It works properly. Now, what I would like to do is to configure it so it plays a notification saying where the call is going to be forwarded. After reading some tutorials, I tried to get it working with rtpproxy, but after configuring it, it doesn't play the clips. I am testing it with a *.wav file. The path to the file is "/usr/local/etc/kamailio/test.wav" (it's in the kamailio directory). This is my software: Debian stable kernel 2.6.32-5-686 Kamailio 3.2.0 (GIT installation) rtpproxy 1.2.1-1 Kamailio and rtpproxy are running in the same machine. These are the log messages: -------------------kamailio.log------------------- (SEVERAL TIMES) Nov 16 12:58:10 debian-virtualbox kamailio[2076]: INFO: rtpproxy [rtpproxy.c:1415]: rtp proxy <udp:127.0.0.1:22222> found, support for it enabled (SEVERAL TIMES) Nov 16 12:58:12 debian-virtualbox kamailio[2069]: ERROR: <script>: Intentando reproducir audio -------------------------------------------------- This is the relevant configuration: --------------/etc/default/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 127.0.0.1" -------------------------------------------------- -----/usr/local/etc/kamailio/kamailio.cfg (just my important changes for this action)--------- loadmodule "rtpproxy.so" modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") request_route { rtpproxy_offer(); xlog("L_ERR","Intentando reproducir audio"); rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav","1"); (...default routes ommited...) route(RELAY); } -------------------------------------------------- I can play it with mplayer, so it's not a sound problem (or even a system codec problem). I would like to know if I can make it working with rtpproxy, or if there is an easier way to do it. Thanks for your time. (I include now the whole kamailio.cfg, just in case you need to check anything) ---------/usr/local/etc/kamailio/kamailio.cfg (complete) ---------- #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # ##-- # WITH_DEBUG # *** To enable mysql: # - define WITH_MYSQL # ##-- #!define WITH_MYSQL # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # ##-- #!define WITH_AUTH # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # ##-- #WITH_NAT # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### 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 ######### #!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 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* 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="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #$banana_server="192.168.0.70" #$treasure_server="192.168.0.60" #$drake_server="192.168.0.60" #!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 ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!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 #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif ##-- loadmodule "sdpops.so" loadmodule "textopsx.so" loadmodule "rtpproxy.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 "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- ##-- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- 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", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!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 { #start_recording(); #force_rtp_proxy(); #rtpproxy_offer(); #rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav", "1"); #("fichero","numero de veces"); #formato alaw? #force_rtp_proxy(); rtpproxy_offer(); xlog("L_ERR","Intentando reproducir audio"); rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav","1"); if ($rU=~"^001788[0-3].*") { xlog("L_ERR", "Banana island"); $rd="192.168.0.70"; msg_apply_changes(); xlog("L_ERR","Redireccionando a $rd\n"); } if ($rU=~"^001788[4-5].*") { xlog("L_ERR", "Big treasure island"); $rd="192.168.0.60"; msg_apply_changes(); xlog("L_ERR","Redireccionando a $rd\n"); } if ($rU=~"^001788[6-7].*") { xlog("L_ERR", "Drake island"); $rd="192.168.0.60"; msg_apply_changes(); xlog("L_ERR","Redireccionando a $rd\n"); } if ($rU=~"^001788[8-9].*") { xlog("L_ERR", "Lost island"); sdp_keep_codecs_by_name("GSM"); #if para si true, bien, y si false mensaje y sale } # 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); ##-- #rtpproxy_offer(); #rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav", "1"); 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 } --------------------------------------------------------------

13 years

Re: [SR-Users] mtree on db Berkeley DB?

by Javier Gallart

> > Hello > > coming back to the topic related to mtree, to be able to set values via > mi/rpc -- it won't be that difficult to add such functionality. Usually > with a tree is mainly reading, due to fast matching on tree indexing. > The question is how many times/often do you need to change values and > how many of them at the same time (more or less). > > I was also thinking about that: our application works in such a way that the full tree is recalculated every 15 minutes. Currently we have aound 60 branches with 150000 entries each. > I assume many times the changes will be somewhere down the tree, since > the first part of the number is usually the same (e.g., country code and > operator prefix). To update the tree at runtime, while there are reads > on it, there must be used a lock to be safe an consistent. If you do lot > of writes and very often, then you keep the tree structure locked a lot, > slowing the search. > That's the case, tname is rarely updated, it's tvalue the column that we normally update. > > Can you estimate the number of writes and how often they would be on a > daily basis? There might be other solutions to look at, if writes are > very often. > >From the numbers above, let's say that we need to update around 8M records every 15 minutes (we expect this number will keep steadily increasing...) As a side note, I've looked at how to implement the mt_match equivalent in redis and it does't look that hard using kamailio s.prefixed transformation (as you suggested) and redis sorted sets. I'll need to make more tests to check the performance, I'll share the results. Thanks! Javi > > Cheers, > Daniel > > > > Regards > > > > Javi >

13 years

Re: [SR-Users] Problem with NAT and RTPProxy

by David Candamil Santos

Hi, I'm just using it because after reading lots of manuals and tutorials, it was the solution I thought it could work. The smartphone is outside the network. I am trying to use it over the Vodafone network. What do you suggest then? Thanks. 2011/11/14 Pavel Segeč <Pavel.Segec(a)fri.uniza.sk> > Hi, **** > > ** ** > > from your topology description I'm not sure why you are trying to use > RTPproxy when all your equipments are on private segments. RTPproxy should > have an public IP address**** > > ** ** > > palo**** > > >

13 years

mtree on db Berkeley DB?

by Javier Gallart

Hi list we've been happily using the mtree module for months now. Lately the size of the tree has grown a lot. The mtree table needs to be fully repopulated and reloaded several times a day, and we are looking for a fastest mechanism (for populating the table, I guess the reload time does not depend much on the db backend...). Does anyone tried with Berkeley DB? Is this combination mtree-berkeley actually feasible...? Thanks Javi

13 years

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sr-users November 2011