- sr-users - kamailio.org

Registration to Kamailio World 2014 is open
by Daniel-Constantin Mierla 14 Jan '14

14 Jan '14

Hello, I would like to announce that the registration to Kamailio World 2014 is now open. This year there is also a pre-conference event for half a day on April 2, with technical tutorials for those looking on more practical and hands-on content. You can see more details on the web site at: - http://conference.kamailio.com/k02/registration/ - http://conference.kamailio.com/k02/2014/01/registration-is-open/ Looking forward to meeting many of you in Berlin during April 2-4, 2014. Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Help with authenticating using Kamailio
by Kasinath 14 Jan '14

14 Jan '14

Hi All, I just installed Kamailio in one server and Asterisk in another. Asterisk loads it sipusers info from database which is in Kamailio server. I don't know how to go further. How can I authenticate Asterisk users through Kamailio. I am trying to authenticate using a sipphone. But no luck. I am missing alot here. I know we can add users using the following command >> kamctl add username password But I already have users in asterisk realtime db. Is there any difference? Awaiting your reply, Thanks in advance, -- Thanks & Regards, Kasinath K P System Engineer Admin-Ahead Server Technologies email:kasinathk@admin-ahead.com

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WebSocket with MSRP
by mohammed alyaseen 14 Jan '14

14 Jan '14

Hi All, I am using Kamailio 4.1 as SIP proxy and registrar with websocket and msrp modules. As a client I am using the JsSIP stack. The client can open websocket for SIP messages by calling "new WebSocket('ws://1.2.3.4:8000','sip')" with sip as a subprotocol, this works fine. But when the client tries to open a websocket for MSRP sessions using new WebSocket('ws://1.2.3.4:8000','msrp') in the browser log I see the error 'WebSocket connection to 'ws://1.2.3.4:8000/' failed: Unexpected response code: 400' . For MSRP file transfer I am using the crocodile msrp lib. The HTTP GET request of sent by the client is: GET / HTTP/1.1 Via: SIP/2.0/TCP 10.147.66.197:54818 Upgrade: websocket Connection: Upgrade Host: 192.168.144.48:8000 Origin: http://localhost Sec-WebSocket-Protocol: msrp Pragma: no-cache Cache-Control: no-cache Sec-WebSocket-Key: gccXstTCf+egnPtufU2xng== Sec-WebSocket-Version: 13 Sec-WebSocket-Extensions: x-webkit-deflate-frame User-Agent: Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/31.0.1650.63 Safari/537.36 Cookie: PHPSESSID=pdlbnc3h24i5r8rdg1979tino2 In the log of Kamailio I see these lines: 11(2114) WARNING: websocket [ws_handshake.c:318]: ws_handle_handshake(): required headers not present 11(2114) DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) 11(2114) DEBUG: <core> [msg_translator.c:204]: check_via_address(): check_via_address(10.147.66.197, 10.147.66.197, 0) 11(2114) DEBUG: <core> [tcp_main.c:2320]: tcpconn_send_put(): tcp_send: send from reader (2114 (11)), reusing fd 11(2114) DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... 11(2114) DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f37ab162bd0 n=182 fd=8 11(2114) DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf= HTTP/1.1 400 Bad Request Via: SIP/2.0/TCP 10.147.66.197:54744 Sec-WebSocket-Protocol: sip Sec-WebSocket-Version: 13 Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 So, anyone has an idea whats the problem is? Thanks, Medo

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Update R-URI after lookup("location")
by Igor Potjevlesch 13 Jan '14

13 Jan '14

Hello, I do an association between X aliases with 1 contact. This contact is connected one time in the location. But sometimes, for some scenario, this contact can be connected 2 or 3 times and the INVITE are sent with parallel forking. When an INVITE is received, after lookup("aliases"), I set the R-URI with the original SIP TO with the following instruction: avp_pushto("$ruri/username", "$tU"); before relay. The issue occurred with the next contacts. They don't passed through this instruction. How can I update all available R-URI in location before relay? Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

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Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Jason Penton 13 Jan '14

13 Jan '14

Hi, I have some questions for you as we have used suspend/continue quite a lot in the IMS code and don't have any leaks. Firstly, why are you using pkg_mem for your hash_id and label? Remember that you will be in 2 different processes in the suspend and continue portions of the code... so pkg_mem will not work - you should use shm_mem instead. Secondly, how are you using top to tell that you have a leak? Kamailio's memory is internally managed. Cheers Jason On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > Re-sending without the attachment. > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, January 13, 2014 4:57 PM > *To:* 'sr-users(a)lists.sip-router.org' > *Subject:* Regd. t_suspend() and t_continue() > > > > Hi, > > > > We are trying out the t_suspend() and t_continue() in our test setup. We > are facing memory leak ( both shm and pkg as per top command results). > > > > Please find below the scenario, > > > > 1) Do a t_newtran() > > 2) Allocate pkg memory for hashid and label. > > 3) Call t_suspend() > > 4) Do t_continue() when async result is available > > 5) De-allocate pkg memory reserved for hashid and label > > 6) Do a t_relay() which forwards the sip message to another sip node. > > > > In the step (6) above, we see t_newtran() allocates one more time shared > memory for the same transaction. > > > > We tried t_release() after step (4) to release the transaction as > t_relay() anyways allocates new shared memory. Nothing helped. > > > > Please let me know what are the logs you would require to debug the same. > I am attaching syslog for this run. > > > > Regards, > > Shankar > > > > > > _______________________________________________ > 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 > > -- *Jason Penton**Senior Manager: Applications and Services**Smile Communications Pty (Ltd)**Mobile:*+27 (0) 83 283 7000*Skype:* jason.barry.pentonjason.penton(a)smilecoms.com <name.surname(a)smilecoms.com> www.smilecoms.com This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

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FW: Regd. t_suspend() and t_continue()
by Shankar 13 Jan '14

13 Jan '14

Re-sending without the attachment. From: Shankar [mailto:shankar.rk@plintron.com] Sent: Monday, January 13, 2014 4:57 PM To: 'sr-users(a)lists.sip-router.org' Subject: Regd. t_suspend() and t_continue() Hi, We are trying out the t_suspend() and t_continue() in our test setup. We are facing memory leak ( both shm and pkg as per top command results). Please find below the scenario, 1) Do a t_newtran() 2) Allocate pkg memory for hashid and label. 3) Call t_suspend() 4) Do t_continue() when async result is available 5) De-allocate pkg memory reserved for hashid and label 6) Do a t_relay() which forwards the sip message to another sip node. In the step (6) above, we see t_newtran() allocates one more time shared memory for the same transaction. We tried t_release() after step (4) to release the transaction as t_relay() anyways allocates new shared memory. Nothing helped. Please let me know what are the logs you would require to debug the same. I am attaching syslog for this run. Regards, Shankar

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Regd. t_suspend() and t_continue()
by Shankar 13 Jan '14

13 Jan '14

Hi, We are trying out the t_suspend() and t_continue() in our test setup. We are facing memory leak ( both shm and pkg as per top command results). Please find below the scenario, 1) Do a t_newtran() 2) Allocate pkg memory for hashid and label. 3) Call t_suspend() 4) Do t_continue() when async result is available 5) De-allocate pkg memory reserved for hashid and label 6) Do a t_relay() which forwards the sip message to another sip node. In the step (6) above, we see t_newtran() allocates one more time shared memory for the same transaction. We tried t_release() after step (4) to release the transaction as t_relay() anyways allocates new shared memory. Nothing helped. Please let me know what are the logs you would require to debug the same. I am attaching syslog for this run. Regards, Shankar

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Critical dialog - Bogus event 5 Problem - PRACK after 200OK
by Arik Lustig 13 Jan '14

13 Jan '14

Hello There, We are detecting errors quite often on our kamailio servers log: /usr/local/sbin/kamailio[17527]: CRITICAL: dialog [dlg_hash.c:782]: bogus event 5 in state 4 for dlg 0 x7fe0269a8be0 [1004:10921] with clid '034e8c8f-07daa8c0-001b56a3(a)192.168.210.2' and tags '07daa8c0-33aeaec-t-1' '9304808465080725853 ' After analyzing the issue We found that the root cause for those errors is when kamailio gets PRACK message after 200 OK (CSeq INVITE) in the same CALLID. To mention that those calls are successfuly established. This is due a complecated voice network. As for today we are getting around 100 errors per kamailio server every day. The number will increase to around 1000 errors per day in the near future. Two questions please: 1. Can Kamailio safety deal with those errors with no service effect 2. Is there a way to decrease the level of those errors from CRITICAL to INFO? As for today we are working in debug level 0 so no INFO written is written to the log in general. I will appriciate for any assistant on this topic please. Thanks, Arik.

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The RPC Function in The TM Module
by 无愿无悔 13 Jan '14

13 Jan '14

Hello everyone: I want to know what is the parameter for the RPC function "tm.cancel" in the TM module ? I input the parameters such as : tm.cancel d90be24a4553d076@Yml5cC5oYWlnzs5uZXQ 2 but it can accept the only parameter,and inform me "error 400-Callid and CSeq expected as parameters " . The context of the Callid.s is "d90be24a4553d076@Yml5cC5oYWlnzs5uZXQ". But the context of the CSeq.s is "if you get this string, you don't check rpc_scan return code!!!(very bad)". Could you give me an example ?

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JSCommunicator feedback?
by Daniel Pocock 10 Jan '14

10 Jan '14

Hi, I'm just wondering if anybody has used JSCommunicator with Kamailio yet and if anybody has any feedback? http://jscommunicator.org/ If anybody publishes a dedicated blog about using them together I'll be happy to link to it I'm also looking for any feedback from anybody who may be able to take the CSS a little bit further. On the one hand, it would be nice to make it look a little bit more slick but on the other hand, the intention is for other people to drop it into their own sites where they have existing styles as well, so I'd prefer not to have it themed so aggressively that users have more work to do removing things. Regards, Daniel

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Older Sip Phone ia hangup after 60sec with Realtime asterisk
by Noriyuki Hayashi 10 Jan '14

10 Jan '14

Hello, I am beginner using kamailio with much appreciated. Only one sip-phone is hang up after 60 seconds problem. This sip phone has no nat function at all.(SANYO SIP-2100) Grand Stream is works fine with kamailio. I would like give me your great advice with much appreciated. Environment. CentOS5.10, Asterisk-11.6.0 with PostgreSQL-9.2.5 as Realtime. Kamailio-4.1.0 Only Asterisk and PostgreSQL with older sip phone works fine. If Kamailio is running that registered is OK, But meetme(example) is hangup after 60 sec. I do not know "reINVITE or RTP" problem. Kamailio.cfg #!KAMAILIO #!enable postgresql #!define WITH_PGSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_NAT #!define WITH_DISPATCHER #!define WITH_ANTIFLOOD #!define WITH_MULTIDOMAIN #!define WITH_WITHINDLG #!define WITH_DEBUG #!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_PGSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "postgres://postgres:password@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "postgres://asterisk:password@localhost/asterisk" #!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 ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 #log_facility=LOG_LOCAL0 log_facility=LOG_LOCAL7 fork=yes children=4 check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes 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 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 */ #listen=udp:192.168.192.92 port=5060 mhomed=1 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!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 #!ifdef WITH_ASTERISK asterisk1.bindip = "192.168.192.92" desc "Asterisk IP Address" asterisk1.bindport = "5080" desc "Asterisk Port" asterisk2.bindip = "192.168.192.93" desc "Asterisk IP Address" asterisk2.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "192.168.192.92" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib64/kamailio/modules/" #!endif #!ifdef WITH_PGSQL loadmodule "db_postgres.so" #!endif 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 #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif #!ifdef WITH_DISPATCHER loadmodule "dispatcher.so" loadmodule "sqlops.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) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # 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", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "name") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- 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) # 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", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif #!ifdef WITH_DISPATCHER # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("sqlops","sqlcon", "ca=>postgres://asterisk:password@localhost/kamailio") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif 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 #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio route(TOASTERISK); exit; } #!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 #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sip_devices", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($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 $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if(ds_is_from_list()) { return 1; } else { return -1; } } # Send to Asterisk route[TOASTERISK] { if($(au{s.len})<=5) { $var(setid) = 0; xlog("SCRIPT: Connected Asterisk #0 - using set $var(setid) \n"); } else { $var(setid) = 9; xlog("SCRIPT: Connected Asterisk #9 - using set $var(setid) \n"); } # failover dispatching on set determined above if(!ds_select_dst($var(setid), "8")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } if($(au{s.len})<=5) { $var(rip) = $sel(cfg_get.asterisk1.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk1.bindport); } else { $var(rip) = $sel(cfg_get.asterisk2.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk2.bindport); } $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } # 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_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } #!endif *** Test call to meetme Logs. **** sip1*CLI> sip set debug on sip1*CLI> SIP Debugging re-enabled sip1*CLI> sip set debug on sip1*CLI> Name/username Host Dyn Forcerport ACL Port Status Description Realtime 99206/99206 192.168.192.92 D N 5060 OK (515 ms) Cached RT 1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline] sip1*CLI> -- Executing [901@99:1] Answer("SIP/99206-00000000", "") Audio is at 15506 sip1*CLI> Adding codec 100003 (ulaw) to SDP sip1*CLI> Adding codec 100008 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP sip1*CLI> <--- Reliably Transmitting (NAT) to 192.168.192.92:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.538bb4a63719b3291fac2770ec3b5b31.0 Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29932-3be9 Record-Route: <sip:192.168.192.92;lr=on;ftag=bec0a8c0-13c4-52cad076-29850-7422;nat=yes> From: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 To: <sip:901@192.168.192.92>;tag=as7cd1f3fc Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 2 INVITE Session-Expires: 120;refresher=uas Contact: <sip:901@192.168.192.92:5080> Content-Type: application/sdp Require: timer Content-Length: 284 v=0 o=root 729993436 729993436 IN IP4 192.168.192.92 s=Asterisk PBX 11.6.0 c=IN IP4 192.168.192.92 t=0 0 m=audio 15506 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> sip1*CLI> Retransmitting #1 (NAT) to 192.168.192.92:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.538bb4a63719b3291fac2770ec3b5b31.0 Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29932-3be9 Record-Route: <sip:192.168.192.92;lr=on;ftag=bec0a8c0-13c4-52cad076-29850-7422;nat=yes> From: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 To: <sip:901@192.168.192.92>;tag=as7cd1f3fc Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 2 INVITE Session-Expires: 120;refresher=uas Contact: <sip:901@192.168.192.92:5080> Content-Type: application/sdp Require: timer Content-Length: 284 v=0 o=root 729993436 729993436 IN IP4 192.168.192.92 s=Asterisk PBX 11.6.0 c=IN IP4 192.168.192.92 t=0 0 m=audio 15506 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- sip1*CLI> <--- SIP read from UDP:192.168.192.92:5060 ---> ACK sip:901@192.168.192.92:5080 SIP/2.0 From: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 To: <sip:901@192.168.192.92>;tag=as7cd1f3fc Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 2 ACK Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.9f229e8f27e7aa8e3fad6cc83135f434.0 Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29a6e-306e Max-Forwards: 16 Contact: <sip:99206@192.168.192.190:5060> Proxy-Authorization: Digest username="99206", realm="192.168.192.92", nonce="UspTFFLKUehXxuLrJ9AbLwT69Jtg5MFQ", uri="sip:901@192.168.192.92", response="8d49c10f184381801aac43fc45b15117", algorithm=MD5 Content-Length:0 <-------------> --- (11 headers 0 lines) --- sip1*CLI> <--- SIP read from UDP:192.168.192.92:5060 ---> ACK sip:901@192.168.192.92:5080 SIP/2.0 From: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 To: <sip:901@192.168.192.92>;tag=as7cd1f3fc Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 2 ACK Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.9f229e8f27e7aa8e3fad6cc83135f434.0 Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29a6e-306e Max-Forwards: 16 Contact: <sip:99206@192.168.192.190:5060> Proxy-Authorization: Digest username="99206", realm="192.168.192.92", nonce="UspTFFLKUehXxuLrJ9AbLwT69Jtg5MFQ", uri="sip:901@192.168.192.92", response="8d49c10f184381801aac43fc45b15117", algorithm=MD5 Content-Length:0 <-------------> --- (11 headers 0 lines) --- sip1*CLI> -- Executing [901@99:2] Wait("SIP/99206-00000000", "1") sip1*CLI> > 0x17aa0bd0 -- Probation passed - setting RTP source address to 192.168.192.190:17096 sip1*CLI> -- Executing [901@99:3] Authenticate("SIP/99206-00000000", "5963") sip1*CLI> -- <SIP/99206-00000000> Playing 'agent-pass.gsm' (language 'ja') sip1*CLI> -- <SIP/99206-00000000> Playing 'auth-thankyou.gsm' (language 'ja') sip1*CLI> -- Executing [901@99:4] MeetMe("SIP/99206-00000000", "99901,pM") == Parsing '/etc/asterisk/meetme.conf': Found sip1*CLI> -- Created MeetMe conference 1023 for conference '99901' sip1*CLI> -- <SIP/99206-00000000> Playing 'conf-onlyperson.gsm' (language 'ja') sip1*CLI> -- Started music on hold, class 'default', on SIP/99206-00000000 sip1*CLI> -- Stopped music on hold on SIP/99206-00000000 sip1*CLI> -- Started music on hold, class 'default', on SIP/99206-00000000 sip1*CLI> Audio is at 15506 Adding codec 100003 (ulaw) to SDP Adding codec 100008 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.192.92:5060: INVITE sip:99206@192.168.192.190:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.92:5080;branch=z9hG4bK71165cd1;rport Max-Forwards: 70 From: <sip:901@192.168.192.92>;tag=as7cd1f3fc To: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 Contact: <sip:901@192.168.192.92:5080> Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.6.0 Session-Expires: 120;refresher=uac Min-SE: 90 Allow: INVITE, ACK, CANCEL, BYE X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 284 v=0 o=root 729993436 729993436 IN IP4 192.168.192.92 s=Asterisk PBX 11.6.0 c=IN IP4 192.168.192.92 t=0 0 m=audio 15506 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- sip1*CLI> <--- SIP read from UDP:192.168.192.92:5060 ---> SIP/2.0 404 Not here Via: SIP/2.0/UDP 192.168.192.92:5080;branch=z9hG4bK71165cd1;rport=5080 From: <sip:901@192.168.192.92>;tag=as7cd1f3fc To: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 102 INVITE Server: kamailio (4.1.0 (x86_64/linux)) Content-Length: 0 --- > [INSERT INTO cdr ("calldate","clid","src","dst","dcontext","channel","lastapp","lastdata","duration","billsec","disposition","amaflags","accountcode","uniqueid") VALUES ('2014-01-06 15:49:12','"Richard Nough" <99206>','99206','901','99','SIP/99206-00000000','MeetMe','99901,pM',60,60,'ANSWERED',3,'nori@wats','1388990952.0')] <--- SIP read from UDP:192.168.192.92:5060 ---> SIP/2.0 404 Not here Via: SIP/2.0/UDP 192.168.192.92:5080;branch=z9hG4bK339a9eb8;rport=5080 From: <sip:901@192.168.192.92>;tag=as7cd1f3fc To: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 103 BYE Server: kamailio (4.1.0 (x86_64/linux)) Content-Length: 0 I hope you have a great 2014. Kind regards, Nori

2 4

Max-Forwards severly decreased when invite is relayed
by Grant Bagdasarian 10 Jan '14

10 Jan '14

Hello, Using Kamailio 4.1. I'm having some problems with the Max-Forwards header being decremented by almost 50 when Kamailio relays the INVITE to the next hop. The INVITE is received by Kamailio with the value of 65, but when sent to the next hop it has the value of 16. I'm not altering the max-forwards header value in any way in the config script. At least not explicitly. Are there any known constructs which indirectly affect the Max-Forwards header? 10.0.0.16 is the Kamailio server. U 10.0.0.10:5060 -> 10.0.0.16:5060 INVITE sip:3110123456@10.0.0.16:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.10;rport;branch=z9hG4bKeg5g6U4Dc5HeS. Max-Forwards: 65. From: "31765123456" <sip:31765123456@10.0.0.10>;tag=9Z7avj87Bc2gm. To: <sip:3110123456@10.0.0.16:5060>. Call-ID: e089c971-f3a8-1231-c2bb-000c2941b070. CSeq: 54291565 INVITE. Contact: <sip:mod_sofia@10.0.0.10:5060>. User-Agent: CM Carrier SBC. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 205. U 10.0.0.16:5060 -> 10.0.0.31:5060 INVITE sip:3110123456@10.0.0.25:5060 SIP/2.0. Record-Route: <sip:10.0.0.16;lr=on>. Via: SIP/2.0/UDP 10.0.0.16;branch=z9hG4bKe501.5459c149e5fa2c6c723149661017aa3d.0. Via: SIP/2.0/UDP 10.0.0.10;rport=5060;branch=z9hG4bKeg5g6U4Dc5HeS. Max-Forwards: 16. From: "31765123456" <sip:31765123456@10.0.0.10>;tag=9Z7avj87Bc2gm. To: <sip:3110123456@10.0.0.16:5060>. Call-ID: e089c971-f3a8-1231-c2bb-000c2941b070. CSeq: 54291565 INVITE. Contact: <sip:mod_sofia@10.0.0.10:5060>. User-Agent: CM Carrier SBC. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 205.

2 2

attrs field in "kamctl fifo ds_list"
by Benjamin Henrion 09 Jan '14

09 Jan '14

Hi, Any idea what attrs are in dispatcher.cfg? When I do a kamctl fifo ds_list, I can put some strings in this field. Is it used for labelling of the SIP URIs? The documentation is a bit obscure for me. If I put hello as the last item on dispatcher.cfg, I see it when I do a ds_list: $ kamctl fifo ds_list SET:: 99 URI:: sip:127.0.0.1:5060 flags=AP priority=0 attrs=hello Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-4148403 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

2 1

Kamailio v4.1.1 Released
by Daniel-Constantin Mierla 09 Jan '14

09 Jan '14

Hello, Kamailio SIP Server v4.1.1 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.0. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.0. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.1. For more details about version 4.1.1 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/01/kamailio-v4-1-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Releasing v4.1.1
by Daniel-Constantin Mierla 09 Jan '14

09 Jan '14

Hello, I am considering to release v4.1.1 out of branch 4.1 this week on Thursday -- there were couple of fixes since we did v4.1.0. If there are any open issues that should get addresses, reply here. Also, for developers, if there are patches only on master branch that should be pushed on 4.1 as well, do it as soon as possible. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 2

Kamdbctl create issue
by Owais ul Haq 09 Jan '14

09 Jan '14

Hello all, I am installing Kamailio-4.1.0 on my system. I get this error while creating 'kamdbctl' file. [root@ss3 sbin]# ./kamdbctl create ERROR: DATABASE ENGINE NOT SPECIFIED, PLEASE SETUP ONE IN THE CONFIG SCRIPT. Can some one please help in solving this error. Any help will be appreciated. Cheers, Kamilio Newbie.

2 1

Kamailio-4.1.0 Installation Issues
by Owais ul Haq 09 Jan '14

09 Jan '14

Hello all, I am installing Kamailio-4.1.0 on my system. I get this error while creating 'kamdbctl' file. Can some one please help in solving this error. Any help will be appreciated. Cheers, Kamilio Newbie.

2 1

question about nat_uac_test
by Brian Davis 08 Jan '14

08 Jan '14

When using nat_uac_test("3"), there is a particular REGISTER message that my kamailio server receives where the source and via addresses are different, but this function seems to return false (and then no nat correction is applied). I have included the register message below. Is there an obvious reason this is not triggering the nat_uac_test? Or can anyone share how they would usually debug a situation like this? Any help/advice would be appreciated. Thank you! Example captured Register message: REGISTER sip:test1.test.com:5060 SIP/2.0 Call-ID: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx CSeq: 1418 REGISTER From: "Joe" <sip:xxxyyyzzzz@test1.test.com>;tag=2163936191 To: "Joe" <sip:xxxyyyzzzz@test1.test.com> Via: SIP/2.0/UDP 96.xxx.xxx.xxx:33745;rport;branch=z9hG4bKf5s1p`n3TRv5TZx5RXy.RVv+JPz8Nat*UX!8KRx4SRx Via: SIP/2.0/UDP 192.xx.xxx.xxx:33745;branch=z9hG4bKeb263246c44095f072d8167dd0c7987a343134;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: "Joe" <sip:xxxyyyzzzz@192.xx.xxx.xxx:33745;transport=udp> Expires: 3600 Authorization: Digest username="xxxyyyzzzz",realm="test1.test.com ",nonce="UsDqxVLA6ZlvIrO3TsqCtm9X3vqY8405",uri="sip:test1.test.com:5060 ",response="769d1e6f2fd36cf040503f9f079279a0" Content-Length: 0 Associated lines from kamailio log: [kamailio]# grep 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxxkamailio.log Dec 30 03:33:45 sip-01 kamailio[20489]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|start|recieved UDP request REGISTER sip:test1.test.com Dec 30 03:33:45 sip-01 kamailio[20489]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|source 96.xxx.xxx.xxx:33745 Dec 30 03:33:45 sip-01 kamailio[20489]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|from sip:xxxyyyzzzz@test1.test.com Dec 30 03:33:45 sip-01 kamailio[20489]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|to sip:xxxyyyzzzz@test1.test.com Dec 30 03:33:45 sip-01 kamailio[20489]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|originated from external sources Dec 30 03:33:45 sip-01 kamailio[20489]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|end|issued new auth challenge to new registration attempt Dec 30 03:33:45 sip-01 kamailio[20495]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|start|recieved UDP request REGISTER sip:test1.test.com:5060 Dec 30 03:33:45 sip-01 kamailio[20495]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|source 96.xxx.xxx.xxx:33745 Dec 30 03:33:45 sip-01 kamailio[20495]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|from sip:xxxyyyzzzz@test1.test.com Dec 30 03:33:45 sip-01 kamailio[20495]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|to sip:xxxyyyzzzz@test1.test.com Dec 30 03:33:45 sip-01 kamailio[20495]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|originated from external sources Dec 30 03:33:45 sip-01 kamailio[20495]: DEBUG: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|authenticated xxxyyyzzzz(a)test1.test.com via cached SIP creds

2 3

Kamailio 4.1.x Installation
by Owais ul Haq 08 Jan '14

08 Jan '14

Hello all, I am currently using Kamailio-3.1.1 installed on a fedora machine. I want to install updated version Kamailio-4.1.x. Can someone give me some link where I can find about installation steps and pre-requisites etc. Thanks for helping. Regards, Kamailio Newbie :)

3 2

kamailio crash after sqlops failures
by Uri Shacked 08 Jan '14

08 Jan '14

Hi kamailio version 3.2.0 16M configured has private memory and 4G as shared memory. using sqlops to query the DB. the DB was down and kamailio crashed after some time with the following log under 1. /var/log/messages "kernel: kamailio[PID]: segfault at 5 ip 00007fdacc65e175 sp 00007fff85897810 error 4 in tm.so[7fdacc630000+79000]" 2./var/log/kamailio " /usr/local/sbin/kamailio[12033]: ERROR: <core> [db_query.c:156]: error while submitting query /usr/local/sbin/kamailio[12033]: ERROR: sqlops [sql_api.c:580]: cannot do the query /usr/local/sbin/kamailio[12055]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 14 /usr/local/sbin/kamailio[11966]: ALERT: <core> [main.c:785]: child process 12034 exited by a signal 11 /usr/local/sbin/kamailio[11966]: ALERT: <core> [main.c:788]: core was generated" (core was not generated due to lack of permissions on linux..... but now it is ok) So, anyone has an idea where to start? BR, Uri

2 4

gateway authentication with the lcr module
by Rene Montilva 08 Jan '14

08 Jan '14

Hi List is there any option with lcr module to authenticate gateways with digest method for outbound calls?, or whats the best practice for that scenario. i have a sip user from my sip provider, it register with uac module for incoming calls, but i want outbound calls too Best Regards

3 2

Re: [SR-Users] Kamailio 4.1, Debian Squeeze, RTPProxy 1.2.1 Segfault [SOLVED]
by Jr Richardson 08 Jan '14

08 Jan '14

On Mon, Dec 30, 2013 at 6:08 PM, Jr Richardson <jmr.richardson(a)gmail.com> wrote: >> >> Hello, >> >> you should enable core dumping - that should be via running 'ulimit -c >> unlimited' before you start rtpproxy. >> >> If you get the coredump, then grab the backtrace with gdb, it should >> show where rtpproxy crashed. With that info, maybe we can spot a fix for it. >> >> Hopefully rtpproxy package is with debug symbols, if not, the look for a >> rtpproxy-dbg package and install it if found. >> >> Cheers, >> Daniel >> >> On 20/12/13 18:59, Jr Richardson wrote: >>> Hi All, >>> root@sip-router3-ve206:/etc/kamailio# kamailio -V >>> version: kamailio 4.1.0 (x86_64/linux) 350d2e >>> Doing some testing, can't seem to get rtpproxy to not segfault. I've >>> loaded version from deb-squeeze pkg, from source >>> http://b2bua.org/chrome/site/rtpproxy-1.2.1.tar.gz and from >>> git://sippy.git.sourceforge.net/gitroot/sippy/rtpproxy >>> <http://sippy.git.sourceforge.net/gitroot/sippy/rtpproxy> (which >>> should be the latest) but all versions segfault as soon as a call sets up. >>> I've configured each versions control socket for both udp or unix, >>> kamailio starts and sees the rtpproxy fine with no errors, but when a >>> call hits, rtpproxy segfaults. >>> I've run rtpproxy in the forground with degug and get this response >>> (IP's washed x.x.x): >>> ------------------ >>> DBUG:handle_command: received command "7505_12 USIEc0,101 >>> 0adb2f8449b8c9026f993a0a7db9ab5d(a)x.x.x.76 x.x.x.76 23388 as33dd7c98;1" >>> INFO:handle_command: new session >>> 0adb2f8449b8c9026f993a0a7db9ab5d(a)x.x.x.76, tag as33dd7c98;1 requested, >>> type strong >>> Segmentation fault >>> ------------------ >>> root@sip-router3-ve206:/etc/kamailio# more /etc/default/rtpproxy >>> # Defaults for rtpproxy >>> # The control socket. >>> CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" >>> # To listen on an UDP socket, uncomment this line: >>> #CONTROL_SOCK="udp:127.0.0.1:7722 <http://127.0.0.1:7722>" >>> # Additional options that are passed to the daemon. >>> EXTRA_OPTS="-l x.x.x.20" >>> kamailio.cfg: >>> loadmodule "rtpproxy.so" >>> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722 >>> <http://127.0.0.1:7722>") >>> inside route[] >>> >>> add_path_received(); >>> rtpproxy_manage("cwei"); >>> record_route(); >>> Any guidance on further identifying the issue? >>> Thanks. >>> JR > > I'm not very familiar with gdb, but did manage to get a core dump and > here are the results: > ------------------------------------------- > root@rtpproxy:/usr/src/rtpproxy-git# gdb > /usr/src/rtpproxy-git/rtpproxy -c /usr/src/rtpproxy-git/core > GNU gdb (GDB) 7.0.1-debian > Copyright (C) 2009 Free Software Foundation, Inc. > License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> > This is free software: you are free to change and redistribute it. > There is NO WARRANTY, to the extent permitted by law. Type "show copying" > and "show warranty" for details. > This GDB was configured as "x86_64-linux-gnu". > For bug reporting instructions, please see: > <http://www.gnu.org/software/gdb/bugs/>... > Reading symbols from /usr/src/rtpproxy-git/rtpproxy...done. > [New Thread 11256] > warning: Can't read pathname for load map: Input/output error. > Reading symbols from /lib/libpthread.so.0...(no debugging symbols found)...done. > Loaded symbols for /lib/libpthread.so.0 > Reading symbols from /lib/libm.so.6...(no debugging symbols found)...done. > Loaded symbols for /lib/libm.so.6 > Reading symbols from /lib/libc.so.6...(no debugging symbols found)...done. > Loaded symbols for /lib/libc.so.6 > Reading symbols from /lib64/ld-linux-x86-64.so.2...(no debugging > symbols found)...done. > Loaded symbols for /lib64/ld-linux-x86-64.so.2 > Reading symbols from /lib/libnss_compat.so.2...(no debugging symbols > found)...done. > Loaded symbols for /lib/libnss_compat.so.2 > Reading symbols from /lib/libnsl.so.1...(no debugging symbols found)...done. > Loaded symbols for /lib/libnsl.so.1 > Reading symbols from /lib/libnss_nis.so.2...(no debugging symbols found)...done. > Loaded symbols for /lib/libnss_nis.so.2 > Reading symbols from /lib/libnss_files.so.2...(no debugging symbols > found)...done. > Loaded symbols for /lib/libnss_files.so.2 > Core was generated by `./rtpproxy -f -l x.x.x.x -s udp:x.x.x.x 7722 -p > /var/run/rtpproxy/rtp'. > Program terminated with signal 11, Segmentation fault. > #0 create_twinlistener (cf=0x7fff957ec2d0, ia=0x0, port=<value > optimized out>, fds=<value optimized out>) > at rtpp_command.c:97 > 97 fds[i] = socket(ia->sa_family, SOCK_DGRAM, 0); > (gdb) bt > #0 create_twinlistener (cf=0x7fff957ec2d0, ia=0x0, port=<value > optimized out>, fds=<value optimized out>) > at rtpp_command.c:97 > #1 create_listener (cf=0x7fff957ec2d0, ia=0x0, port=<value optimized > out>, fds=<value optimized out>) > at rtpp_command.c:143 > #2 0x000000000040967d in handle_command (cf=0x7fff957ec2d0, > controlfd=<value optimized out>, > cmd=0x7f73bf148d00, dtime=<value optimized out>) at rtpp_command.c:837 > #3 0x000000000040bc9d in process_commands (cf=0x7fff957ec2d0, > controlfd_in=3, dtime=<value optimized out>) > at rtpp_command_async.c:63 > #4 0x000000000040bd75 in rtpp_cmd_queue_run (arg=<value optimized > out>) at rtpp_command_async.c:92 > #5 0x00007f73bff6e8ca in start_thread () from /lib/libpthread.so.0 > #6 0x00007f73bfa53b6d in clone () from /lib/libc.so.6 > #7 0x0000000000000000 in ?? () > (gdb) > --------------------------------------------------------------------------------- > I could not find an rtpproxy debug package and the install notes do > not tell how to compile with debugging symbols. > > Thanks. > > JR So apparently rtpproxy doesn't like to be told to use 2 interfaces in the flags (ie or ei) when only 1 interface is actually enabled for use. This was a case of RTFM! RTPProxy seems to be working fine, once I got the correct config parameters in kamailio. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Cassandra module
by Carlos Ruiz Díaz 08 Jan '14

08 Jan '14

Hi, does anybody know if it is possible to use Kamailio's built-in maintenance tools with the db schema as tables in Cassandra? I'm trying to create the "subscriber" table in a Cassandra cluster and maintain it using kamctl options for new users and passwords. I tried modifying the DBENGINE in kamctlrc to DB_CASSANDRA, CASSANDRA, DBCASSANDRA, without luck. Regards, -- Carlos http://caruizdiaz.com +595981146623

2 1

tag column on address table
by Kelvin Chua 07 Jan '14

07 Jan '14

is the peer_tag_avp used on allow_source_address_group? Kelvin Chua

3 11

SIP Security Architectural Question to Use RTP/Media Proxy or Not?
by Jr Richardson 07 Jan '14

07 Jan '14

Hi All, Background: We are a service provider offering VoIP/Data services to business customers. All hosted VoIP systems and Customers are mostly on-net, VoIP systems not exposed to the Internet, but all hosted PBX's do have public IP address. I do have some Customers with off-net phones/users so I basically white list their IP's so the phones can register back to their hosted PBX. This works well and keeps SIP attack vectors to a minimum. I've been working on a single point of registration Kamailio server to backend PBX's so I can further control public Internet access to hosted PBX's. I've got this working in the lab but have some concerns about RTP streams. I know I can use a RTP/Media Proxy to also have a single point of entry for media streams to the the backend PBX's but don't believe this to be the best method. Researching SBC's and what I know about SIP and RTP Streams, it's best to have media controlled via the B2BUA (Asterisk in this case) and since all my hosted PBX's have public IP's there would be no compelling reason to proxy RTP adding another hop, latency and point of failure other than for security. I'm not transcoding media or doing anything outside of the capability of the B2BUA as far as media goes. Question: Would it be prudent to open UDP media ports from Internet to PBX's on a case-by-case basis, basically white listing media streams or is there any attack vulnerability with UDP in the media port range or should I open up media port range to all PBX's and not worry about attacks. Are there any UDP Media exploits that I should be concerned with, or UDP flood attacks that could DOS my hosted PBX's? Thanks for any feedback. JR -- JR Richardson Engineering for the Masses

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