January 2014 - sr-users

Kamailio not mdifying Contact in multi-homed setup
by Biswas, Dipak 31 Jan '14

31 Jan '14

Hi All, I'm running into an issue, I'm not sure whether any of you seen this yourselves and resolved it. Please share some pointers. My network is: clients <--> Public IP(Kamailio/RTPProxy)10.1.128.11 <--> 10.1.128.34 (Freeswitch) The 200 OK response from Freeswitch (on the way back from called party to caller) to Kamailio is shown below. Notice the Contact header URI host part contains Freeswitch Private IP (10.1.128.34). Kamailio suppose to change that to Public IP before forwarding the 200 OK (copied below) to Caller in public domain. But, it's not. As a result, ACK from Caller is not reaching back to Kamailio. How did you or anybody out there using Kamailio resolve this problem? If needed, I can copy/paste my kamailio.cfg. SIP 200 OK -> Via: SIP/2.0/UDP 10.1.128.11;branch=z9hG4bKa7ea.4013d9881c1b7fe7b4c6c0f0e8f9d6b6.0 Via: SIP/2.0/UDP <Caller Public IP>:5060;rport=5060;branch=z9hG4bK-383736-9b4621118533d3ccea17992738433249 Record-Route: <sip:10.1.128.11;r2=on;lr=on;ftag=791b5ae3;nat=yes> Record-Route: <sip:<Kamailio Pub IP>;r2=on;lr=on;ftag=791b5ae3;nat=yes> From: \"Dipak Biswas\" <sip:dipak.biswas@<Kamailio Pub IP>>;tag=791b5ae3 To: <sip:dipak.biswas.test2@<Kamailio Pub IP>>;tag=atXF5443gQj9p Call-ID: 20d5d6d366fcb06c259a0895b3e44b52@0:0:0:0:0:0:0:0 CSeq: 2 INVITE Contact: <sip:dipak.biswas.test2@10.1.128.34:5060;transport=udp> User-Agent: FreeSWITCH mod_sofia/1.4.2+git~20140108T200418Z~d8fc8469b4~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 269 -- Thanks, Dipak

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sip reroute question
by Øystein Rudi 31 Jan '14

31 Jan '14

Hi, I have a scenario where multiple companies with multiple sip and smtp domains want to also use one common domain. Ie. Users from domain1.com have their own Exchange and Lync topology, this is also true for users from domain domain2.com Is there a way to use SER on commondomain.com to forward Lync traffic, presence, chat, rtp to these servers/users B.r, Øystein Rudi Teknisk leder (CTO) / Seljestadveien 78 A, 9407 Harstad sentralbord: +47 77 00 27 50 / fax: +47 77 00 27 51 / web: www.serit.no epost: oystein.rudi(a)itpartner.no / direkte: +47 77 00 27 81 / mobil: +47 93 00 89 99 [Serit_ITPartner_Harstad]

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File transfering problem with RTPproxy
by Wingsravi R 31 Jan '14

31 Jan '14

Dear Daniel & Kamailio'ns I am working on File transferring feature between two SIP clients (IMSDroid). I have configured Kamailio (4.0.4) sever with RTPproxy to get the through NAT traversal of audio/videov calls. so fro that i added a script in kamailio.cfg as like below (Forced all INVITES through RTPproxy) : if (is_method("INVITE")) { setflag(7); # Set the qos flag #if (has_sdp()) { if(has_body("application/sdp")) { if (rtpproxy_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } #if (is_method("ACK") && has_sdp()) if (is_method("ACK") && has_body("application/sdp")) rtpproxy_answer(); .. .. onreply_route[1] { # if (has_sdp()) if(has_body("application/sdp")) rtpproxy_answer(); } onreply_route[2] { # if (has_sdp()) if(has_body("application/sdp")) rtpproxy_offer(); } But now when i am trying for File transferring (As per RFC 4975, Session mode IM) their INVITEs are also passing through RTPProxy service, which is dropping MSRP contents and with that Files are not getting transferring (But session is establishing between two clients). And when i tried file transferring without running RTPPproxy service. files are getting transferring. but that gives problems for audio/video calls. So anybody please help me in how to make it work for both MSRP&RTP packets. I mean how can i modify the script so that INVITE messges contains MSRP-RTP contents can be descriminate in the script, and then they can execute their respective blocks of scripts. And please find the attachments of my Kamailio.cfg file and file transfer traffic capture (tcpdump based) with RTPproxy instance running and without running. Any help will greatly appreciate. Awaiting for somebody's help. Regards Ravi

3 5

401 responses ending up in onreply_route instead of failure_route
by Daniel Tryba 31 Jan '14

31 Jan '14

Situation: endpoint -> kamailio (4.1.0) -> upstream sip server (kamailio 4.0.x) For an outbound call endpoint authenticates with kamailio. Kamailio relays the invite to upstream and upstream responds with a 401. This 401 should trigger authentication. So I thought I could use uac_auth() to fix this. uac_rely may only be used from FAILURE_ROUTE. So the initial INVITE from the endpoint has this set: if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } But to my suprise the 401 doesn't end up in the failure_route but in the onreply_route. Why? What am I missing? -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

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Re: [SR-Users] Kamailio without tls
by Neville D'Souza 31 Jan '14

31 Jan '14

And also let me know wat can cause voice break up during conversation. Our Server has an public IP but our clients are behind nat. From: Neville D'Souza [mailto:ncd@gogopay.com] Sent: Friday, January 31, 2014 5:45 PM To: 'sr-users(a)lists.sip-router.org' Subject: Kamailio without tls Hi Support, I have recently kamailio 4.0.x version with tls on port 5061. Now our development team needs to connect to it using unencrypted on port 5060 to test an app for features and then work towards connecting through tls connection. So my question is how do I temporarily disable tls to test all the features on the kamailio server and then once feature testing is done revert back to tls connection? Best Regards, Neville D'Souza

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one audio
by bkyeyune 31 Jan '14

31 Jan '14

Hello;iam running kamailio 4 with two interfaces one public and the other privateam using the default kamailio.cfg file on ubuntu*scenario*register and route calls on asterisk through the proxy private interface on the lan interfaceregister and route calls on asterisk through the public interface if you are on the internetbe able to call from either lan or internet to any one registered on the lan or internet*Achieved *can make register and make calls between endponts both on the internet two way audiocalls from the internet endpoint to lan registered end point one way audiocalls from lan endpoint to internet endpoint no audiocalls from lan to lan endpoints no audio*help*configuration of rtpproxy rule to allow audio flow to all endpointslan to lan lan to internet internet to lanusing the default configuration file route -n0.0.0.0 public 0.0.0.0 UG 1 0 0 eth00.0.0.0 private 0.0.0.0 UG 2 0 0 eth1public 0.0.0.0 pub sunet U 0 0 0 eth0private 0.0.0.0 pri subnet U 0 0 0 eth1/etc/default/rtpproxyCONTROL_SOCK="-F -s udp:*:7722"EXTRA_OPTS="-l public/private -d DBUG:LOG_LOCAL0 -u rtpproxy" -- View this message in context: http://sip-router.1086192.n5.nabble.com/one-audio-tp124687.html Sent from the Users mailing list archive at Nabble.com.

1 0

Notify with Websocket problem
by David Claybourne 31 Jan '14

31 Jan '14

Hi Kamailio-Community, i have the following problem. Publish and subscribe works as suspected. But after saving the presence state from a publish request and notifying the users via NOTIFY, users with a websocket connection don't get the notify request at all :(. Here is the version of kamailio i use: version: kamailio 4.1.1 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. I created a log file and attached the notify event (the problem is that a websocket dont have a real IP Address), is there any possibility to use the nathelper module for fixing this problem? I also attached the config file. Thanks for the help Greetings David Log: 9(25950) DEBUG: presence [notify.c:1470]: send_notify_request(): dialog info: 9(25950) DEBUG: presence [notify.c:117]: printf_subs(): pres_uri: sip:1001@example.com 9(25950) DEBUG: presence [notify.c:118]: printf_subs(): watcher_user@watcher_domain: 1000(a)example.com 9(25950) DEBUG: presence [notify.c:119]: printf_subs(): to_user@to_domain: 1001(a)example.com 9(25950) DEBUG: presence [notify.c:120]: printf_subs(): from_user@from_domain: 1000(a)example.com 9(25950) DEBUG: presence [notify.c:121]: printf_subs(): callid/from_tag/to_tag: sc19bmkpmluh91btkc4t7s(a)example.com/o0cqavak8u/6853f4be2e83d57dcdf85008ad4e4098-9f19 9(25950) DEBUG: presence [notify.c:122]: printf_subs(): local_cseq/remote_cseq: 2/4 9(25950) DEBUG: presence [notify.c:123]: printf_subs(): local_contact/contact: sip:5.45.101.128:8088;transport=ws/sip:1000@7b1g33n1ji.invalid 9(25950) DEBUG: presence [notify.c:124]: printf_subs(): record_route: 9(25950) DEBUG: presence [notify.c:125]: printf_subs(): sockinfo_str: tcp:5.45.101.128:8088 9(25950) DEBUG: presence [notify.c:126]: printf_subs(): event: presence 9(25950) DEBUG: presence [notify.c:127]: printf_subs(): status: active 9(25950) DEBUG: presence [notify.c:128]: printf_subs(): reason: 9(25950) DEBUG: presence [notify.c:129]: printf_subs(): version: 2 9(25950) DEBUG: presence [notify.c:130]: printf_subs(): expires: 595 9(25950) DEBUG: presence [notify.c:132]: printf_subs(): updated/updated_winfo: 0/0 9(25950) DEBUG: presence [notify.c:1563]: send_notify_request(): headers: Max-Forwards: 70 Event: presence Contact: <sip:5.45.101.128:8088;transport=ws> Subscription-State: active;expires=595 Content-Type: application/pidf+xml 9(25950) DEBUG: presence [notify.c:925]: ps_build_dlg_t(): CONTACT = sip:1000@7b1g33n1ji.invalid 9(25950) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==12 && [5.45.101.128] == [5.45.101.128] 9(25950) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 8088 (advertise 0) matches port 8088 9(25950) DEBUG: presence [notify.c:1573]: send_notify_request(): expires 595 status 1 9(25950) DEBUG: presence [notify.c:1726]: shm_dup_cbparam(): === 22/8/37 9(25950) DEBUG: tm [uac.c:243]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sip:1000@7b1g33n1ji.invalid> 9(25950) DEBUG: <core> [dns_cache.c:569]: _dns_hash_find(): dns_hash_find((0), 33), h=0 9(25950) DEBUG: <core> [dns_cache.c:569]: _dns_hash_find(): dns_hash_find(7b1g33n1ji.invalid(18), 1), h=745 9(25950) ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su: could not resolve hostname: "7b1g33n1ji.invalid" 9(25950) ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "7b1g33n1ji.invalid" 9(25950) ERROR: tm [uac.c:266]: t_uac_prepare(): t_uac: no socket found 9(25950) ERROR: presence [notify.c:1590]: send_notify_request(): in function tmb.t_request_within 9(25950) ERROR: presence [notify.c:1677]: notify(): sending Notify not successful 9(25950) ERROR: presence [notify.c:1275]: publ_notify(): Could not send notify for presence Config: #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.0 - 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/wiki/ # 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 # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** 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 # # *** 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 ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### #LOAD MODULES NEEDED #!define WITH_AUTH #!define WITH_PRESENCE #!define WITH_MYSQL #!define WITH_TLS #!define WITH_WEBSOCKETS #!define WITH_DEBUG ##!define WITH_MSRP #!substdef "!MY_IP_ADDR!5.45.101.128!g" #!substdef "!MY_DOMAIN!5.45.101.128!g" #!substdef "!MY_WS_PORT!8088!g" #!substdef "!MY_WSS_PORT!8089!g" #!substdef "!MY_MSRP_PORT!9000!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" #!substdef "!MY_MSRP_ADDR!tls:MY_IP_ADDR:MY_MSRP_PORT!g" #!substdef "!MSRP_MIN_EXPIRES!1800!g" #!substdef "!MSRP_MAX_EXPIRES!3600!g" # *** 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. #!ifndef DBURL #!define DBURL "mysql://kamailio:databasepassword@localhost/kamailio" #!endif #!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 ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!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="example.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 #!ifdef WITH_TLS enable_tls=yes #!endif #listen=MY_IP_ADDR #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR #!ifdef WITH_TLS listen=MY_WSS_ADDR #!endif #!endif #!ifdef WITH_MSRP listen=MY_MSRP_ADDR #!endif tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 #syn_branch=0 # 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" pstn.gw_port = "" desc "PSTN GW Port" #!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/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.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_MSRP loadmodule "msrp.so" loadmodule "htable.so" loadmodule "cfgutils.so" #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "nathelper.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 --------------- # ----- 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", 120) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$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_WEBSOCKETS # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is _only_ being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. #!endif #!ifdef WITH_MSRP # ----- htable params ----- modparam("htable", "htable", "msrp=>size=8;autoexpire=MSRP_MAX_EXPIRES;") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else { if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } #!endif # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) 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] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) 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()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } 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 route(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_SPEEDDIAL # 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); } route(RELAY); exit; } # 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 #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && 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 { add_contact_alias(); } 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)) { add_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!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}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($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 $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) || $Rp == MY_MSRP_PORT) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; exit; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif #!ifdef WITH_MSRP event_route[msrp:frame-in] { msrp_reply_flags("1"); if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) && $Rp != MY_MSRP_PORT) { xlog("L_WARN", "MSRP request received on $Rp\n"); msrp_reply("403", "Action-not-allowed"); exit; } if (msrp_is_reply()) { msrp_relay(); } else if($msrp(method)=="AUTH") { if($msrp(nexthops)>0) { msrp_relay(); exit; } if (!www_authenticate("MY_DOMAIN", "subscriber", "$msrp(method)")) { if (auth_get_www_authenticate("MY_DOMAIN", "1", "$var(wauth)")) { msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Server Error"); } exit; } if ($hdr(Expires) != $null) { $var(expires) = (int) $hdr(Expires); if ($var(expires) < MSRP_MIN_EXPIRES) { msrp_reply("423", "Interval Out-of-Bounds", "Min-Expires: MSRP_MIN_EXPIRES\r\n"); exit; } else if ($var(expires) > MSRP_MAX_EXPIRES) { msrp_reply("423", "Interval Out-of-Bounds", "Max-Expires: MSRP_MAX_EXPIRES\r\n"); exit; } } else { $var(expires) = MSRP_MAX_EXPIRES; } $var(cnt) = $var(cnt) + 1; pv_printf("$var(sessid)", "s.$(pp).$(var(cnt)).$(RANDOM)"); $sht(msrp=>$var(sessid)::srcaddr) = $msrp(srcaddr); $sht(msrp=>$var(sessid)::srcsock) = $msrp(srcsock); $shtex(msrp=>$var(sessid)) = $var(expires) + 5; # - Use-Path: the MSRP address for server + session id $var(hdrs) = "Use-Path: msrps://MY_IP_ADDR:MY_MSRP_PORT/" + $var(sessid) + ";tcp\r\n" + "Expires: " + $var(expires) + "\r\n"; msrp_reply("200", "OK", "$var(hdrs)"); } else if ($msrp(method)=="SEND" || $msrp(method)=="REPORT") { if ($msrp(nexthops)>1) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay(); exit; } $var(sessid) = $msrp(sessid); if ($sht(msrp=>$var(sessid)::srcaddr) == $null) { # one more hop, but we don't have address in htable msrp_reply("481", "Session-does-not-exist"); exit; } else if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay_flags("1"); msrp_set_dst("$sht(msrp=>$var(sessid)::srcaddr)", "$sht(msrp=>$var(sessid)::srcsock)"); msrp_relay(); } else { msrp_reply("501", "Request-method-not-understood"); } } #!endif

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t_relay_to_tls cannot be found
by jaflong jaflong 31 Jan '14

31 Jan '14

Can anyone suggest why t_relay_to_tls cannot be found 0(15284) ERROR: <core> [cfg.y:3272]: yyparse(): cfg. parser: failed to find command t_relay_to_tls i am using kamailio 4.1.1

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Kamailio is not starting when it connects against remote mysql server
by José Seabra 30 Jan '14

30 Jan '14

Hi there, I'm having a strange behavior when i start kamailio, it start processes but after few seconds all processes die, <core> [main.c:1630]: main_loop(): main_loop: Cannot fork Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7354]: DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening connection: mysql:// xxxx:xxxx@10.0.30.15/kamailio Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Can't connect to MySQL server on '10.0.30.15' (4) Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: uri_db [checks.c:249]: uridb_db_init(): unable to connect to the database Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: <core> [sr_module.c:896]: init_mod_child(): init_mod_child(): Error while initializing module uri_db (/usr/local/lib64/kamailio/modules/uri_db.so) Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: <core> [pt.c:350]: fork_process(): ERROR: fork_process(): init_child failed for process 5, pid 7335, "udp receiver child=4 sock=10.0.20.7:5060 ( 213.58.213.226:5060)" Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: : <core> [main.c:1630]: main_loop(): main_loop: Cannot fork anyone has idea why it is happening, it was working before and we didn't any changes -- Cumprimentos José Seabra

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Re: [SR-Users] Kamailio behind NAT
by John Smith 30 Jan '14

30 Jan '14

Hello Klaus, I have been making some tests just to be sure the network is not the problem. While using a simpler config file for Kamailio with just the TLS and NAT which does not involve Asterisk in the scenario, calls work properly between softphones. OTOH, using the template provided in the KB referenced in my first post disabling the Asterisk define, calls are connected properly but no audio flows between phones. Now I am in the process of trying to locate where is the problem by comparing how both files handle the NAT support. Thank you > ----- Original Message ----- > From: Klaus Darilion > Sent: 01/23/14 08:12 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > On 23.01.2014 10:29, John Smith wrote: > > Hello Klaus, > > > > I had already two sockets bound each to two independent physical interfaces. I have added the force_send_socket at each rtpproxy > > Just for clarity: > > force_send_socket is for near_end NAT traversal of the SIP signaling, > whereas manage_rtpproxy() is for the NAT traversal (near end and far > end) of the RTP stream. > > > It is necessary to use the cwie / cwei flags in the rtpproxy_manage call? > > If rtpproxy uses only a single listen-IP, then these flags are not > needed. Only if you operate rtpproxy in bridge mode, then you need these > flags. Bridge mode is necessary if you do not have IP routing between > the internal network and the "virtual external" network, or if you want > to bridge between IPv4 and IPv6. > > > Currently audio does not flow back to the softphones, it gets lost at Kamailio. > > Actually it should get lost at rtpproxy. > > Please post a SIP trace: ngrep -Wbyline -q -t -P "" port 5060 > > and post the setup (external + internal IP addresses) (you can send it > privately to me or mangle the IP addresses if they are sensitive) > > regards > Klaus > > _______________________________________________ > 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

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