- sr-users - kamailio.org

dialog timeout reinvite
by Kelvin Chua 09 Jan '15

09 Jan '15

Happy new year everyone! :) I have stumbled into a case where a switch would send a reinvite to my kamailio server every 15 minutes. I am setting dialog timeout using avp for prepaid purposes. now i think it resets the timer upon processing of the reinvite. what is the best way to work around this?

3 5

listening to specific interface with ip
by Jijo 09 Jan '15

09 Jan '15

HI All. I know that kamailio support listen on an interface with port (listen = eth0:5060). Does it support listen on an interface with ip, port, transport? For example listen=eth0: udp:10.10.10.10:5060 Thanks Jijo

2 1

Can you use the sr.modf() method / function to call append_branch?
by J Lee 08 Jan '15

08 Jan '15

I would like to know if there's a way to do parallel forking via lua code. So specifically, I'd like to be able to do something like this: seturi("sip:a@example.com"); append_branch("sip:b@example.com"); append_branch("sip:c@example.com"); append_branch("sip:d@example.com"); t_relay(); except via lua, because the list of branches is dynamic. I've tried the following: sr.modf("append_branch","sip:10.x.x.x") but the error message I'm getting is: /usr/sbin/kamailio[7437]: ERROR: app_lua [app_lua_sr.c:177]: lua_sr_modf(): function 'append_branch' is not available Just wondering if I "should" be able to do this. I've been reading http://www.kamailio.org/wiki/embeddedapi/devel/lua#srtm_package but haven't seen any references to append_branch. The other alternative I'm looking at is to just use lua to generate the list of destinations and pass back a hashtable of sorts to the cfg file. Then from there I will try to loop through the hashtable and do the append_branch command for each entry in the table. If you have other suggestions, I'd appreciate it. Thanks.

2 1

Kamailio v4.2.2 Released
by Daniel-Constantin Mierla 08 Jan '15

08 Jan '15

Hello, Kamailio SIP Server v4.2.2 stable release is out. This is a maintenance release of the latest stable branch, 4.2, that includes fixes since release of v4.2.1. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.2. For more details about version 4.2.2 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/01/kamailio-v4-2-2-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

Planning to release v4.2.2
by Daniel-Constantin Mierla 08 Jan '15

08 Jan '15

Hello, I am considering to release next patch update in 4.2 series next week, most probably on Thursday, January 8, 2015. Given the winter holidays, if there are issues not being taking care of, send a reminder to mailing list or, even better (less chances to be forgotten), open a case on Github issue tracker: - https://github.com/kamailio/kamailio/issues Wishing everyone a great 2015! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 7

dispatcher weight dont work
by Yuriy Gorlichenko 08 Jan '15

08 Jan '15

Hello I use dipatcher algorithm that works with weight. I added 2 Asterisks and try to call its with my kam, but this still works like 4 algorithm. Weight does not work. How I must configure dispatchr for working with weight? My configuration now is modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@10.0.1.1") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") modparam("tm", "reparse_on_dns_failover", 0)

2 3

Dialog update timeout
by Pars3c 07 Jan '15

07 Jan '15

Hi, I wish to update dialog timeout when i receive the 200ok in the on_reply route I try to make it with the dlg_set_timeout but it give an error "bogus dialog..." How can i make it? Thanks

4 12

3rd party REGISTER body
by Jyaim Jyaim 07 Jan '15

07 Jan '15

Hello, I'm wondering if I can configure Kamailio to get in 3rd party REGISTER request's body the initial REGISTER request received from user ? Thanks in advance and best regards.

4 6

Hiding topology header From
by Kalala Alexander 07 Jan '15

07 Jan '15

2 2

Queue implementation in kamailio
by Ansuman Manna - Asergis Telecom 07 Jan '15

07 Jan '15

Hi, I am developing a system using Kamailio 4.2.1 as a front-end to a group of Asterisk servers. All of the usual functionality (registations, inbound, outbound, internal calls) are fine. We need to add call queueing to the system. I looked at the mohqueue module and read that we can use mohq_process() to send inbound calls to a suitably defined queue, based on the incoming URI. But think is that how can I get sip extension status in the particular queue? Any help would be appreciated. Thanks. Regards, Ansuman Manna Software Developer Asergis Telecom Services Private Limited Office Tel: +91 11 6606 8400 Office Fax: +91 112554 5156 http://www.asergis.in <http://www.asergis.in>

2 4

Hiding topology header From
by Kalala Alexander 07 Jan '15

07 Jan '15

1 1

Unregister is failing
by Anil Kumar 07 Jan '15

07 Jan '15

Hi All, I am running ims servers(pcscf,scscf,icscf and hss) as part of kamailio proxy. And I am trying to register and unregister the end-points, with help osip lib. But I see the below ERROR message. ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio/pcscf/kamailio.cfg] l=908 a=25 n=t_check_status Please help me to fix the issue. Thanks and Regards, Anil

2 1

No audio when connect from public network but it works for lan users
by CK Lee 06 Jan '15

06 Jan '15

I am new to Kamailio after starting to play around for 3 weeks. Before Kamailio, I used asterisk and it works quite well in my network which has 2 public IP addresses and 10+ wired and wireless users. The asterisk listens to only one IP which my netgear router port forward the necessary sip and rdp ports to the asterisk. My interest is to use Kamailio as SIP proxy to sit in front of the asterisk. I follow the tutorial here - http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I use debian wheezy and kamailio is installed from the repository with version 4.2.1 Everything works for my lan subscribers. However, when I am outside the lan, I can register but do not the incoming packets at my sip client. What I observe is two strange behaviours 1. in asterisk, the kamailio peer has to use "sip.example.org or its public IP" in the host name field, otherwise it is not online 2. in looking at the log, I found the sip client peer is registered at my lan address i.e. 192.168.118.xx instead of a public IP when I directly login to asterisk. I add on rtproxy but not succeed with error as force_rtp_proxy_body: incorrect port 0 in reply from rtpproxy Below is my kamailio.cfg Help is appreciated. -------------------------------------- #!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/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** 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 ####### Defined Values ######### #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_NAT #!define WITH_TLS #!define ADDR_IPV4 192.168.118.30 # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://user: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 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias="sip.example.org" /* 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:192.168.118.30:5060 listen=tls:192.168.118.30:5061 /* 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 #!ifdef WITH_XCAPSRV tcp_accept_no_cl=yes #!endif tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT #tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.118.30" desc "Asterisk IP Address" asterisk.bindport = "15066" desc "Asterisk Port" kamailio.bindip = "192.168.118.30" 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_k:modules" #!else mpath="/usr/lib/i386-linux-gnu/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.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" loadmodule "outbound.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 # ----------------- setting module-specific parameters --------------- modparam("ctl", "user", "kamailio") # ----- 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) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # 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:127.0.0.1:22222") #modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock") # ----- 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", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # 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] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif # 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; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rtpproxy_offer(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # 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] { # 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; } #!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("$fd", "sipusers", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($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 $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.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(); } #!endif

2 4

Kamailio Active / Active HA.
by Mahmoud Ramadan Ali 05 Jan '15

05 Jan '15

Hi Dears, I have successfully configured Kamailio HA using Heartbeat and Pacemaker so if one of the two servers should go down the other server will own the virtual IP address and take over. But i have two questions: 1.This model is considered to be Active / Passive redundancy so one server will process the SIP signaling until it goes down and i'm wondering is there is any way two achieve Active / Active redundancy and if so how the signaling will be handled on the servers so they can be aware of the transactions and dialogs traversing the servers ? does the signalling will be replicated or synchronized between the servers or what ?! 2.What about the servers DB and how they should be designed in a cluster mode ? i want to replicate all the DBs of the server to get consistent user registration using the subscriber table for example. Thanks in advance and Best regards.

7 14

Kamailio fails to generate core dump
by Alex Balashov 03 Jan '15

03 Jan '15

Hi, I recently had a simultaneous crash on three production instances of Kamailio. Unfortunately, I have not been able to get to the bottom of the issue because Kamailio refused to produce a core dump: Dec 30 15:37:29 xxxxxx /usr/local/sbin/kamailio[13743]: ALERT: <core> [main.c:784]: handle_sigs(): child process 13763 exited by a signal 11 Dec 30 15:37:29 xxxxxx /usr/local/sbin/kamailio[13743]: ALERT: <core> [main.c:787]: handle_sigs(): core was not generated I do not have disable_core_dump=yes (default is no), and checked ulimits for limits on core file size. Kamailio was running as root, and there was no limit. What are the other reasons why Kamailio may not generate a core dump in such a case? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

3 5

Double attempt to record-route, RR param not added
by ron.kamailio＠mcleodnet.com 03 Jan '15

03 Jan '15

I have been away from Kamailio for a while and starting working on a project where it would be a good fit. I am having a problem when I try to add a Record-Route header with a tag. I see a message in the log indicating that a Double attempt to record-route was made, and the tag that I specified (;rtpproxy=yes), does not get included. I have checked-over the scripts defined in the kamailio.cfg, and I am very sure that record_route() does not get called more than once. Also, the log only shows record_route being execute a single time. Can anyone suggest how I can troubleshoot this problem further? 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=206 a=57 n=set_adv_addr 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=207 a=58 n=set_adv_port 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=208 a=25 n=record_route_preset 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=209 a=62 n=force_send_socket 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=210 a=2 n=exit 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=235 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=231 a=41 n=isflagset 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=232 a=25 n=add_rr_param 5(32087) DEBUG: rr [record.c:899]: add_rr_param(): rr_param_buf=<;rtpproxy=yes> 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=235 a=25 n=xlog 5(32087) ERROR: <script>: new ru: sip:19999009595@192.168.100.52:5080 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=236 a=24 n=record_route 5(32087) ERROR: rr [rr_mod.c:262]: w_record_route(): Double attempt to record-route 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=237 a=5 n=route 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=247 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=244 a=25 n=is_method 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=246 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=245 a=25 n=t_is_set 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=245 a=25 n=t_on_branch 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=250 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=247 a=25 n=is_method 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=249 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=248 a=25 n=t_is_set 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=248 a=25 n=t_on_reply 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=254 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=250 a=25 n=is_method 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=252 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=251 a=25 n=t_is_set 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=251 a=25 n=t_on_failure 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=257 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=254 a=24 n=t_relay The complete log file, script files, and SIP trace can be found here: http://www.siplabs.net/tracebin/record-route/kamailio-log.txt http://www.siplabs.net/tracebin/record-route/kamailio.cfg http://www.siplabs.net/tracebin/record-route/bgcf-local.cfg http://www.siplabs.net/tracebin/record-route/sip-trace.txt Thanks and Happy New Year, Ron

2 3

presence - Error deleting subscription
by Jan Gaida 03 Jan '15

03 Jan '15

Hello, We are upgrading our servers to Kamailio 4.2 and we see a lot of error messages "Failed to delete subscription from memory". Scenario: This happens every time there is a Notify Timeout (408) and Kamailio tries to delete the subscription. Consequences: I don't know whether the delete fails because the entry did never exist, was already deleted or because the comparison (to_tag, callid or from_tag) is incorrect. It would be nice to at least have some details in the log message. I have not noticed any consequences to general functionality. I suspect the commit (74d73953327ad7702922b27ba37c71e3283659c3) to be responsible for this error because it introduced some extra checks on deletion. Does this happen to anybody else? What may be configured incorrectly on our system? Kind regards *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.gaida(a)grupoamper.com | www.grupoamper.com -- This message and any attachments are intended only for the use of the individual to whom they are addressed and it may contain information that is privileged or confidential. If you have received this communication by mistake, please notify us immediately by e-mail or telephone.The storage, recording, use or disclosure of this e-mail and its attachments by anyone other than the intended recipient is strictly prohibited. This message has been verified using antivirus software; however, the sender is not responsible for any damage to hardware or software resulting from the presence of any virus. Este mensaje y cualquier anexo son exclusivamente para la persona a quien van dirigidos y pueden contener información privilegiada o confidencial. Si usted ha recibido esta comunicación por error, le agradecemos notificarlo de inmediato por esta misma vía o por teléfono. Está prohibida su retención, grabación, utilización o divulgación con cualquier propósito. Este mensaje ha sido verificado con software antivirus; sin embargo, el remitente no se hace responsable en caso de que en éste o en los archivos adjuntos haya presencia de algún virus que pueda generar daños en los equipos o programas del destinatario.

2 4

t_suspend()/t_continue() problem
by Dmytro Bogovych 02 Jan '15

02 Jan '15

Greetings. I'm trying to adopt kamailio to handle incoming calls / generate push notifications for softphone running on Windows Phone 8. Starting point was this publication http://www.kamailio.org/events/2014-KamailioWorld/day2/26-Daniel-Constantin… I adopted the script from publication a bit - now notification URI is extracted from REGISTER messages + INVITE transactions are not resuming when unregistering happens. However it does not work for me yet. I see the suspicious output in kamailio log when softphone registers after incoming push notification. Please ignore ERROR: level name - it is just dirty logging call, not real problem. Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: ERROR: <script>: resuming trasaction [40951:63820139] dbogovych (40951:63820139) Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: tm [t_lookup.c:1635]: t_lookup_ident(): DEBUG: t_lookup_ident: transaction found Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:181]: lookup(): contact for [dbogovych] found by address Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 90211984 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 91061218 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 91229062 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 90750343 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 87612750 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 91841578 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: WARNING: tm [t_lookup.c:1537]: t_unref(): WARNING: script writer didn't release transaction Can this "script writer didn't release transaction" be the problem? Thank you :)

1 2

Happy New Year!
by Daniel-Constantin Mierla 01 Jan '15

01 Jan '15

A very dynamic 2014 for Kamailio has reached its end! Thank you everyone for contributing to it! Looking forward to 2015, a lot of new features in Kamailio and new contributors! I wish a healthy and prosperous year to all Kamailio friends, hoping to meet many of you at Kamailio World Conference and other events around the globe! Happy New Year! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 3

Making media packets associated with calls appear unidirectional using rtpproxy and kamailio
by aft 31 Dec '14

31 Dec '14

Hi, I'm trying use two instances of rtpproxy listening to two different public IPs, to handle incoming and outgoing legs. That means i want to implement this network topology: SipClient--------->media from client to proxy--->rtpproxy1 Sipclient<----------media from proxy to client<---rtpproxy2 Where rtpproxy1 listening to IP1 and rtpproxy2 listening to IP2 in kamailio cfg : modparam("rtpproxy", "rtpproxy_sock", " udp:127.0.0.1:22222 udp: 127.0.0.1:22223") if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } rtpproxy_offer("IP1"); } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } rtpproxy_answer("IP2"); } But its not working. Thanks in advance

1 0

kamailio 4.2.1 snmpstats module depends on libsnmp15
by René Bormann 31 Dec '14

31 Dec '14

hello guys, just wanted to add the snmpstats modules to our kamailio 4.2.1 installation. but we are running into a libssnmp15 dependency. searching google showed me that on 14.04 the lib is called libsnmp30 now. does anybody know how to fix this? thanks, rené ------------------- root@kamailio1:~# apt-get install kamailio-snmpstats-modules Reading package lists... Done Building dependency tree Reading state information... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: kamailio-snmpstats-modules : Depends: libsnmp15 (>= 5.4.3~dfsg) but it is not installable E: Unable to correct problems, you have held broken packages. root@kamailio1:~#

2 1

Dialog_end from event route
by Dan-Cristian Bogos 30 Dec '14

30 Dec '14

Hey Guys I need assistance for the following case: I receive dialog_end request from external component in evapi:message-received block and would like to somehow kill the dialog corresponding to h_id and h_entry I get in the message. Is there any way to do that without involving mi_*? Any tip appreciated! Ta, DanB.

3 2

about kamailio
by ThatOther Dev 30 Dec '14

30 Dec '14

is it possible with ypur server project to call to country mobile networks just from a pc, without special equipment?

2 1

ul.dump failing when more than 26 users are registered
by João Vitor Arruda 30 Dec '14

30 Dec '14

Hello, I´m doing a test with the ul.dump output on both Kamailio 4.1.5 or 4.2.1 and it works great if the number of registered users is 26 or lower. Starting with 27 registered users the "kamcmd ul.dump" command returns the error: "error: 500 - Internal error creating aor struct" In Kamailio 4.2.1 the new (and nice) "kamcmd stats.get_statistics usrloc:" returns the correct number of registered_users without this "26 limit". Its possible to check why ul.dump is failing this way? Thanks,

2 3

'from' user != auth username
by Dima T. 30 Dec '14

30 Dec '14

Hello, I need migrate users from the old proprietary softswitch but they have predefined from/to header which is differ from auth username. I read about flags in auth_check but I need strict pairs from user/auth username. How this authentication scheme could be implemented with Kamailio? -- Dmitry

2 1

kamailio configuration problem
by javier falbo 30 Dec '14

30 Dec '14

Hi, I would like some help from a kamailio expert. I have installed on a Debian7 64 kbytes VPS this tutorial: http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour, Problem is that there is no voice in a zrtp connection, but i hear only the ring sound ok. Strange is that on my local wifi works all perfect including the voice, but with external ip no. Any help? I would like to hire a kamailio expert for fixing this and also adding some extras. Write me directly so we could get in contact. regards, javier

2 2

Re: [SR-Users] bug in allow_source_address?
by Henry Fernandes 29 Dec '14

29 Dec '14

Does anyone have feedback about this? -H From: Henry Fernandes <henry(a)usinternet.com> Date: Wednesday, December 17, 2014 at 3:39 PM To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: bug in allow_source_address? I¹m using allow_source_address in my Kamailio config and have been for months/years. I¹ve had no issues with it until today. The issue is that allow_source_address("1²) returns ³1" (for $si=123.17.29.201 and $sp=5060) even though we don¹t have this IP address in the address¹ table for group 1. We have defined this IP address for group 2 but not group 1. select * from address order by mask; +----+-----+---------------+------+------+----------+ | id | grp | ip_addr | mask | port | tag | +----+-----+---------------+------+------+----------+ | 20 | 1 | 123.17.0.0 | 23 | 5061 | Testing | | 22 | 1 | 123.17.0.0 | 23 | 5062 | Testing | | 23 | 1 | 123.17.0.0 | 23 | 5063 | Testing | | 24 | 1 | 123.17.0.0 | 23 | 5064 | Testing | | 1 | 1 | 123.17.0.0 | 23 | 5060 | | | 4 | 1 | 20.5.176.0 | 24 | 5060 | | | 21 | 1 | 123.17.1.235 | 32 | 5060 | Override | | 46 | 2 | 123.17.29.201 | 32 | 5060 | customer | +----+-----+---------------+------+------+----------+ Can anyone explain this? I¹m wondering if there¹s a bug in how allow_source_address does its net mask calculations. -H

2 2

about destination blacklisting
by Juha Heinanen 29 Dec '14

29 Dec '14

i made a test where domain name in request uri hostpart has two udp srv records, one with higher priority than the other. when i turn off the higher priority uac, there is about 5 sec delay before the lower priority uac starts to ring. if i make another call within a small number of seconds from the previous one, my sip proxy remembers that the higher priority uac is dead and the lower priority one starts to ring immediately. but if i wait about five seconds or more, my sip proxy again first tries the higher priority uac and there is again the about 5 second delay before the lower priority one starts to ring. i have in config use_dst_blacklist=on dst_blacklist_expire=20 from the docs i get the impression that my sip proxy should remember 20 seconds that the higher priority uac is dead, but it only remembers it about 5 seconds. what could be the explanation? -- juha

1 0

RTPProxy metod UPDATE
by Kalala Alexander 29 Dec '14

29 Dec '14

1 0

What's happening? Follow us on more channels than e-mail!
by Olle E. Johansson 29 Dec '14

29 Dec '14

Friends, The Kamailio project has great mailing lists for discussions, but did you know that we also spread news in various other channels? If you use these channels, please follow us there and spread the word about Kamailio! Facebook https://www.facebook.com/Kamailio Twitter @kamailioproject Google plus has both a Kamailio page and a discussion group - Page https://plus.google.com/u/0/b/114984922071338017631/114984922071338017631/p… - Group https://plus.google.com/communities/104535904708990386927 In addition to the news from the web site there are a few additional tips and tricks coming on these channels. See you there! /O

1 0

SIPURI sending private IP address
by Cibin Paul 28 Dec '14

28 Dec '14

Hi, I have a kamailio 4.1 as a gateway and registrar behind asterisk servers. On asterisk, I save the IP address of the originating call parsing the SIPURI or SIPCHANNEL(recvip). In both cases, I am receiving the private ip of the user agent registered with kamailio. Do I need to change anything in kamailio to receive the actual IP address. Please suggest. Regards Cibin

3 5

Meddling with password during authentication
by Olli Heiskanen 27 Dec '14

27 Dec '14

Hello all, During authentication, is there any way to affect the password user is sending? I do suspect not as it is a clear security matter, but won't hurt to ask. I use auth_db module with calculate_ha1 parameter set to 1. For reasons in integrating Kamailio into my system architecture there is a need to store a password in some other format than for example md5('555:domain.com:password)') while not allowing any passwords to be stored as plaintext. For example: md5('555:domain.com:md5('password')') but this would require me to hash the password before authentication, in Kamailio script as I can't do it in the clients. Reason for this question is to have my users in a separate database, and these users could have 0-n sip peers assigned to them, and have users authenticate to my software and the sip peers using the same password. cheers, Olli

3 3

Authorization Issue kamailio 4.2
by Ahmed Salem 27 Dec '14

27 Dec '14

Hi, I'm using kamailio 4.2 some of my clients are getting 401 unauthorized after a couple of days of usage without any problems, their usernames and passwords are still in the database server unchanged. Any idea what's going on? Thanks Ahmed

3 2

Merry Christmas!
by Daniel-Constantin Mierla 25 Dec '14

25 Dec '14

Another year getting slowly to the end, an excellent opportunity to thank to all old and new friends of Kamailio project, from contributing code to using it, helping on mailing lists and other places online, or advertising it across the world! Merry Christmas and Happy Winter Holidays! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 3

Media trouble with kamailio/rtpengine
by Marc Soda 24 Dec '14

24 Dec '14

I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not getting audio back to my browser. From a packet capture I can see media from the browser to rtpengine, and then bi-directional RTP back and forth from my asterisk server, but rtpengine is not sending the media on to the browser, i.e.: browser ---------> kamailio/rtpengine <---------> asterisk This is the output from rtpengine: https://gist.github.com/marcantonio/bfe72644306b205cc7e1 Thanks.

2 4

Dispatcher weight nor working
by Yuriy Gorlichenko 24 Dec '14

24 Dec '14

Hello. We use kamailio 4.3 and dispatcher with 9 algorithm. We use db instanse for dispatcher and at attrs column for our backend servers set WEIGHT=40 and 60 for first and second server, but packets sended only at first server ignoring weight

1 0

Re: [SR-Users] Mediaproxy do not work well with Kamailio SIP server issue
by Alfred C.M. Lu 24 Dec '14

24 Dec '14

|---------+-----------------------------> | | dong-ming.li(a)mail.| | | foxconn.com | | | | | | 2014/12/24 上午 | | | 11:54 | |---------+-----------------------------> >--------------------------------------------------------------------------------------------------------------------------------------------------| | | | To: alfred.lu(a)foxconn.com | | cc: | | Subject: Mediaproxy do not work well with Kamailio SIP server issue | >--------------------------------------------------------------------------------------------------------------------------------------------------| Secret Level: |------------------------| | ( ) 1.Non-Confidential | | ( ) 2.Confidential | | ( ) 3.Secret | | ( ) 4.Top Secret | |------------------------| Priority Level: |--------------------| | ( ) 1.Normal | | ( ) 2.Urgent | | ( ) 3.Extra Urgent | |--------------------| ------------------------------------------------------------------------- Hi, sir, We try to setup Mediaproxy on CentOS 6.6 with kamailio-4.2.1 SIP server. We use Linphone android version as SIP client to test. attached kamailio.cfg.txt is our kamailio-4.2.1 SIP server config file. pls help to rename kamailio.cfg.txt to kamailio.cfg. kamailio-4.2.1 SIP server is run on public IP server,and Linphone SIP client is run behind NAT. The problem is that only callee can hear caller's rtp audio data. caller can not hear callee's audio data. And the call will auto disconnect after about 30s time out. Below's our modification regarding the kamailio.cfg file,pls help to check if this is OK or not ? #foxconn gallice add start #### MediaProxy module loadmodule "mediaproxy.so" modparam("mediaproxy", "disable", 0) modparam("mediaproxy", "ice_candidate", "high-priority") modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy/dispatcher.sock") modparam("mediaproxy", "mediaproxy_timeout", 500) modparam("mediaproxy", "signaling_ip_avp", "$avp(nat_ip)") modparam("mediaproxy", "media_relay_avp", "$avp(media_relay)") #foxconn gallice add end # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting use_media_proxy(); #foxconn gallice add } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { #foxconn gallice add #route(NATMANAGE); #foxconn gallice delete use_media_proxy(); #foxconn gallice add } #foxconn gallice add } For mediaproxy,we can see the media related processes are already run. 31212 ? SL 0:01 python ./media-dispatcher start 31218 ? SL 0:16 python ./media-relay start ***************************************************** dong-ming / 李東明 Foxconn Electronics Inc. HH Precision Ind. Co., Ltd. CNSBG NSDI TIPBU R&D Dept. LH Office 地址:深圳市龍華新區龍華東環二路二號富士康科技集團分機 : +86-0755-2812-9588 #26109 Mobile : 13510123670 Mail : dong-ming.li(a)mail.foxconn.com, ***************************************************** 溫馨提醒﹕入廠人員須遵守富士康出入管制,請勿攜帶手提電腦、各種攝像、U盤、MP3 等存儲設備本電子郵件及附件所載信息均為保密信息，受合同保護或依法不得洩漏。其內容僅供指定收件人按限定範圍或特殊目的使用。未經授權者收到此信息均無權閱讀、使用、複製、洩漏或散佈。若您因為誤傳而收到本郵件或者非本郵件之指定收件人，請即刻回覆郵件或致電Super Notes郵件客服熱線 560-104，並永久刪除此郵件及其附件和銷毀所有複印件。謝謝您的合作！ This e-mail message together with any attachments thereto (if any) is confidential, protected under an enforceable non-disclosure agreement, intended only for the use of the named recipient(s) above and may contain information that is privileged, belonging to professional work products or exempt from disclosure under applicable laws.Any unauthorized review, use, copying, disclosure, or distribution of any information contained in or attached to this transmission is STRICTLY PROHIBITED and may be against the laws. If you have received this message in error, or are not the named recipient(s), please immediately notify the sender by e-mail or telephone at Super Notes support hotline 560-104 and delete this e-mail message and any attached documentation from your computer. Receipt by anyone other than the intended recipient(s) is not a waiver of any attorney-client or work product privilege. Thank you! mail from ip-->192.168.2.2 mail from pc-->MICROSOF-700F48 Version: Super Notes 1.6.4.5B (See attached file: kamailio.cfg.txt)

1 0

TLS enable false.
by Thanh Truong 23 Dec '14

23 Dec '14

Hi all, I have tried several configure TLS in kamailio but no luck. Please give me some suggestion that I can make it work correctly. This is my configure in TLS module. modparam("tls", "tls_method", "SSLv23") modparam("tls", "private_key", "/usr/local/etc/kamailio/ca/privkey.pem") modparam("tls", "certificate", "/usr/local/etc/kamailio/ca/kamailio1_cert.pem") modparam("tls", "ca_list", "/usr/local/etc/kamailio/ca/calist.pem") modparam("tls", "verify_certificate", 1) modparam("tls", "require_certificate", 1) I am only getting issue with verify_certifiate = 1, it i let it to 0, my client can register correctly. When I set this flag, i got message from server as: Dec 18 10:26:31 17237 /usr/local/sbin/kamailio[12655]: ERROR: <core> [tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: tls [tls_server.c:1193]: tls_read_f(): TLS accept:error:140890C7:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:peer did not return a certificate Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: <core> [tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading I Cant add any pem file into client, i have used Blink phone but no luck. Thank all in advance. ThanhTruong.

3 3

[BOUNTY] Configs for Video calls between Linphone Android App and WebRTC clients (JsSIP, SIPML)
by Andrey Utkin 23 Dec '14

23 Dec '14

I hope the subject line says it all. I need to set up a backend to establish calls between browser clients and usual SIP clients, and the abovementioned applications are of the highest interest. (Of course, calls between same UAs must work, too.) If Linphone is not going to work due to its internal issues, then other established open source mobile apps for iOS and Android will be fine. Currently I have configured latest kamailio+rtpengine with configs from here https://github.com/caruizdiaz/kamailio-ws The calls pass this way currently: jssip -> android: no sound from phone to the browser, i see that jssip sends "sendonly" attribute for audio in INVITE's SDP. Audio from browser to phone, and both video streams appear immediately, everything is fine with them. android -> jssip: video from browser to the phone appears in 1-2 _minutes_ after the call is answered. All other media streams are fine. The above results are the same in such browsers [IP-] [ ] www-client/firefox-bin-31.3.0:0 [IP-] [ ~] www-client/google-chrome-unstable-41.0.2243.0_p1:0 With sipml currently i have no stable results, so it's hard to describe what happens. Please contact me if you have configs to make the needed things work, or if you have experience of such things working stable, and can configure it quickly. -- Andrey Utkin

2 4

websocket ws_handle_handshake() error
by jaflong jaflong 23 Dec '14

23 Dec '14

Hi list, I am getting this error "ERROR: websocket [ws_handshake.c:153]: ws_handle_handshake(): retrieving connection" Any idea why? This is the debug log 12(24113) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg... 12(24113) DEBUG: xhttp [xhttp_mod.c:358]: xhttp_handler(): new fake msg created (537 bytes): <GET /ws HTTP/1.1 Via: SIP/2.0/TLS 10.1.1.1:45968 Host: 10.1.1.2:10443 User-Agent: Mozilla/5.0 (X11; Linux x86_64; rv:34.0) Gecko/20100101 Firefox/34.0 Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8 Accept-Language: en-US,en;q=0.5 Accept-Encoding: gzip, deflate Sec-WebSocket-Version: 13 Origin: http://mydomain.co.uk Sec-WebSocket-Protocol: sip Sec-WebSocket-Key: GRbV+ovG+DKIBJtwb4pRiQ== Connection: keep-alive, Upgrade Pragma: no-cache Cache-Control: no-cache Upgrade: websocket > 12(24113) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: 12(24113) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <GET> 12(24113) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: </ws> 12(24113) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <HTTP/1.1> 12(24113) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 12(24113) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 12(24113) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via 12(24113) DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) 12(24113) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header 12(24113) DEBUG: <core> [msg_translator.c:205]: check_via_address(): check_via_address(10.1.1.1, 10.1.1.1, 0) 12(24113) DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (24113 (12)), reusing fd 12(24113) DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... 12(24113) DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7fc08162ad68 n=421 fd=7 12(24113) DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf= �/�*�޹��!��S��zo�e�/D��V�Gs��s�� ^cd��S"Ŏ�K��<�c7��4lx�<Z��#�c(AJ�8V��d� ��4bF��Wu�(�w�a� f��n��s�KjQv�݈�q|Z��64{Wk��Z\ױ��M�,[�Arٵ��Q�_��u�d�Gza�f��I�fI��z��Š9=��Ħ3�ȀJPx�}��S?��eVe��\id��y�wP�r(kZs��Z�"��4`�N_MR��z�!��Eb}��+�\;�8�3��V��l��M�|�1�2��VVK�@%�a�B&�P��k 12(24113) INFO: <script>: Received HTTP request to /ws from [10.1.1.1:45968] with protocol tls 14(24115) DEBUG: <core> [tcp_main.c:3565]: handle_ser_child(): handle_ser_child: read response= 7fc08162ad68, -1, fd -1 from 12 (24113) 14(24115) DEBUG: tls [tls_server.c:604]: tls_h_close(): Closing SSL connection 0x7fc081626e20 12(24113) ERROR: websocket [ws_handshake.c:153]: ws_handle_handshake(): retrieving connection 12(24113) DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) 12(24113) DEBUG: <core> [msg_translator.c:205]: check_via_address(): check_via_address(10.1.1.1, 10.1.1.1, 0) 12(24113) ERROR: sl [../../forward.h:247]: msg_send(): tcp_send failed 12(24113) ERROR: websocket [ws_handshake.c:113]: ws_send_reply(): sending reply 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) 12(24113) DEBUG: <core> [receive.c:298]: receive_msg(): receive_msg: cleaning up 12(24113) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0xa31f40, 7, -1, 0x10) fd_no=2 called 12(24113) DEBUG: <core> [tcp_read.c:1434]: release_tcpconn(): releasing con 0x7fc08162ad68, state -2, fd=7, id=3 12(24113) DEBUG: <core> [tcp_read.c:1435]: release_tcpconn(): extra_data 0x7fc081626e20 14(24115) DEBUG: <core> [tcp_main.c:3331]: handle_tcp_child(): handle_tcp_child: reader response= 7fc08162ad68, -2 from 2 9(24110) DEBUG: websocket [ws_conn.c:459]: wsconn_get_list(): wsconn_get_list

1 0

Rpc dialog variable
by Pars3c 22 Dec '14

22 Dec '14

Hi How can i do to see the dialog variables like $dlg_var with RPC ? It's possible? Thanks

1 0

ERROR: dispatcher [dispatch.c:1402]: ds_load_remove(): cannot find load for
by Will Ferrer 22 Dec '14

22 Dec '14

Hi All We recently discovered that our dispatcher boxes have this error constantly at the end of our calls. ERROR: dispatcher [dispatch.c:1402]: ds_load_remove(): cannot find load for (<callid goes here>) We aren't seeing any issues arising from it that we can notice but wanted to see if any one knew what it meant and what it might cause. Thanks much in advance. All the best Will Ferrer Switchsoft Inc

2 7

Kamailio not start if redis server is down
by pars3c 22 Dec '14

22 Dec '14

Hi, i'm trying to use the ndb_redis connector , but i've a problem when redis server is down and i try to restart kamailio. Kamailio in this case not start and release this log: 2(19590) ERROR: ndb_redis [redis_client.c:112]: redisc_init(): error communicating with redis server [srv1] (xxxxxx/0): Connection refused 2(19590) ERROR: ndb_redis [ndb_redis_mod.c:126]: child_init(): failed to initialize redis connections 2(19590) ERROR: <core> [sr_module.c:927]: init_mod_child(): init_mod_child(): Error while initializing module ndb_redis (//lib/kamailio/modules/ndb_redis.so) 30(19622) ERROR: <core> [tcp_main.c:4962]: tcp_init_children(): ERROR: tcp_main: fork failed: Connection refused 31(19624) : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 11 31(19624) : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 31 0(19588) ALERT: <core> [main.c:775]: handle_sigs(): child process 19589 exited normally, status=255 0(19588) ALERT: <core> [main.c:775]: handle_sigs(): child process 19590 exited normally, status=255

2 2

rtpengine stats
by Marc Soda 22 Dec '14

22 Dec '14

Does anyone know how a can get stats from rtpengine? I see the $rtpstat pseudo variable in Kamailio, but from the documentation it looks like that will only give me stats on a particular call. I'm looking for overall stats like concurrent calls, bandwidth, etc... Thanks, Marc

3 2

Multi-tenant SBC Kamailio
by Kalala Alexander 22 Dec '14

22 Dec '14

3 5

Re: [SR-Users] sr-users Digest, Vol 115, Issue 16
by Kees Meijs - SIGNET B.V. 22 Dec '14

22 Dec '14

Hi there, We haven't used gdb on Kamailio yet but some time soon we'll most likely will. Obviously I'll post our findings. Thanks again! Cheers, Kees On 17-12-14 12:00, sr-users-request(a)lists.sip-router.org wrote: > did you get a core dump file? If yes, send the full backtrace taken with > gdb.

2 1

kamailio dispatcher module stops processing REGISTER
by José Seabra 22 Dec '14

22 Dec '14

Hello there, I'm doing stress testing against kamailio (sip load balancer role) sending sip registrar, my stress test software is sending 50 registrations per second, after 30 seconds and 1000 registers processed the function ds_select_dst (I'm using algorithm 4) returns null for all registers sent from my software, and all registrar sip servers are available and in active mode, if I reduce the software to send 20 registrar's per second, the issue only happens after 60 seconds and kamailio (sip load balancer role) process 4000 registrers, after this, function ds_select_dst starts returning null Anyone has any idea why it is happening? My kamailio version is 4.2 Best Regards -- Cumprimentos José Seabra

3 9

The "Subscriber" table VS the "Location" table ?!
by Mahmoud Ramadan Ali 20 Dec '14

20 Dec '14

Hi Dears, I'm a little bit confused about the difference between the "Subscriber" table and the "Location" table.I read that the "Location" table is used for persistent user registration BUT i did NOT configured the "Location" table and can get persistent user registration with the "Subscriber" table ! So can anyone explain the difference please ? Best Regards and Thanks in Advance.

3 2

rtpengine returns 0.0.0.0 even with sendrecv in offer
by Juha Heinanen 19 Dec '14

19 Dec '14

i got mozilla to generate sdp with sendrecv, but still rtpengine does not replace 0.0.0.0 address on o and c lines. why? -- juha Dec 19 10:20:18 box /usr/bin/sip-proxy[5841]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1) Dec 19 10:20:18 box rtpengine[20171]: Got valid command from 127.0.0.1:39586: offer - { "sdp": "v=0#015#012o=Mozilla-SIPUA-34.0 25949 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-ufrag:f5511cb8#015#012a=ice-pwd:1d062e6b56db4f738c7613bc97086194#015#012a=fingerprint:sha-256 45:CA:2E:A5:12:4E:80:0C:AD:9D:11:94:C5:2D:ED:15:55:FD:83:70:91:59:3D:57:3D:F9:A5:77:74:85:BD:E8#015#012m=audio 9 RTP/SAVPF 109 9 0 8 101#015#012c=IN IP4 0.0.0.0#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8 ... Dec 19 10:20:18 box rtpengine[20171]: ... 000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level#015#012a=setup:actpass#015#012a=rtcp-mux#015#012a=candidate:0 1 UDP 2122252543 192.168.1.52 42190 typ host#015#012a=candidate:0 2 UDP 2122252542 192.168.1.52 54801 typ host#015#012a=candidate:1 1 UDP 1686110207 192.98.100.128 42190 typ srflx raddr 192.168.1.52 rport 42190#015#012a=candidate:1 2 UDP 1686110206 192.98.100.128 54801 typ srflx raddr 192.168.1.52 rport 54801#015#012", "ICE": "force", "replace": [ "session-connection", "origin" ], "call-i ... Dec 19 10:20:18 box rtpengine[20171]: ... d": "lr92g1vkbkvmgdt9m1dg", "via-branch": "z9hG4bK2313203", "received-from": [ "IP4", "192.98.100.128" ], "from-tag": "d03ug77pi6", "command": "offer" } Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Creating new call Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Opened ports 8056..8057 for media relay Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Opened ports 8012..8013 for media relay Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Returning to SIP proxy: d3:sdp725:v=0#015#012o=Mozilla-SIPUA-34.0 25949 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-lite#015#012m=audio 8056 RTP/SAVPF 109 9 0 8 101#015#012c=IN IP4 0.0.0.0#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=rtcp:8057#015#012a=rtcp-mux#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8FwYLdKkGvVH4Ykc+kTgrFN1Cwz0KZVpT2JK7+g0#015#012a=setup:actpass#015#012a=fingerprint:sha-1 ED:DE:59:BA:F7 ... Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] ... :F8:78:9D:1D:78:42:AA:00:6C:88:85:A4:2B:36:42#015#012a=ice-ufrag:4dmKpmV3#015#012a=ice-pwd:75jY5mekvVULGZnnB9mJmHQoVa9i#015#012a=candidate:bVSyZZLCFRG94R1D 1 UDP 2130706431 192.26.111.29 8056 typ host#015#012a=candidate:bVSyZZLCFRG94R1D 2 UDP 2130706430 192.26.111.29 8057 typ host#015#0126:result2:oke

2 7

Initial REGISTER request/response in 3rd Party Register
by jyaim 19 Dec '14

19 Dec '14

Hello, I'm using Kamailio with a SIP Application Server, when a user registers on the IMS Core a 3rd Party REGISTER request is sent to the application server to start some logic. I'm trying to add in the 3rd party register request body the initial REGISTER request sent by user device, and the 200 OK response sent back by Kamailio. Do you think it is possible to configure Kamailio to get this behaviour ? Thank you!

1 0

Accounting: syslog vs db_flatfile
by DanB 19 Dec '14

19 Dec '14

Hey Guys, We are in the process to review the interoperability between Kamailio and CGRateS and was wondering what is the best/most recommended/maintained way to generate accounting from Kamailio: * Syslog * db_flatfile * other way maybe? Is CDR generation planned for files already, or just accounting events? Thanks in advance! DanB

6 8

SIP Fragments
by Marc Soda 19 Dec '14

19 Dec '14

I'm having a problem reassembling UDP packets on my Asterisk servers after passing through Kamailio (it appears to me an OS level issue, nothing to do with Kamailio). Is there a way, with Kamailio, to limit the size of a SIP message? I know I can just start removing headers, but that doesn't seem like a realistic solution. I see that Kamailio can compress the message body, but can Asterisk handle that? How do other people handle this? Thanks in advance, Marc

7 13

webrtc clients support using rtpengine
by Andrey Utkin 19 Dec '14

19 Dec '14

Hi! I need to establish calls between WebRTC and usual SIP clients (exactly, sipml/jssip and linphone-android). I used configs from https://github.com/caruizdiaz/kamailio-ws and latest git master HEAD of both kamailio and rtpengine. I got calls from webrtc to android working correctly (but only with Firefox browser), even with video. But in other directions i have some issues because of lack of RTP delivery or RTP timeouts. I have some logs to show you regarding this: https://gist.github.com/krieger-od/27c6f3e4924f5e21352e (works), https://gist.github.com/krieger-od/196bcfbd331d621427ef (doesn't work). I would really love to get some quick help from anyone. For direct manual fixing, I can give a couple of hundreds of bucks. Looking forward impatiently for reply from anyone having something to say. -- Andrey Utkin

3 10

Kamailio World 2015 – Call For Speakers
by Daniel-Constantin Mierla 18 Dec '14

18 Dec '14

Hello, a short note to announce for everyone interested that submissions of presentation proposals is now open for Kamailio World 2015 (Berlin, May 27-29). More details at: - http://conference.kamailio.com/k03/2014/12/kamailio-world-2015-call-for-spe… Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio CDRs
by Massimo Varriale (IPZeta) 18 Dec '14

18 Dec '14

Hi Guys! I'm trying to store CDRs for Kamailio calls. Following Siremis tutorial (http://siremis.asipto.com/install-accounting/) I'm able to gather Succesfull Calls and using the MySQL stored procedure I can have a CDR table. But what I can see now is that only outbound calls with a duration are stored and that failed attempts are not stored into the database. How to capture those attempts too? Here is my accounting configuration: # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd;src_ip=$si") modparam("acc", "cdr_enable", 1) #!endif Thanks Max

2 3

Any extension or solution on how to implement RPC over SIP
by Thierry Luo 18 Dec '14

18 Dec '14

Dear all, I'm trying to design a SIP based framework that can execute RPC end-to-end. Session is set up based on SIP, and then one user agent can remote call a method into the peer user agent. Anyone have any solution on this? Thanks. Best Regards. Thierry Luo

1 0

Maintenance work on kamailio.org - Wed, Dec 17
by Daniel-Constantin Mierla 17 Dec '14

17 Dec '14

Hello, on Wednesday (Dec 17) late afternoon in Europe (probably starting around 18:00, Berlin time), we plan some maintenance work on kamailio.org server. Hopefully the downtime will be short and barely noticeable. But as we know, sometime we have to expect the unexpected, therefore in case the server is unavailable a bit longer, keep in mind that it is a scheduled work. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 2

bug in allow_source_address?
by Henry Fernandes 17 Dec '14

17 Dec '14

I¹m using allow_source_address in my Kamailio config and have been for months/years. I¹ve had no issues with it until today. The issue is that allow_source_address("1²) returns ³1" (for $si=123.17.29.201 and $sp=5060) even though we don¹t have this IP address in the address¹ table for group 1. We have defined this IP address for group 2 but not group 1. select * from address order by mask; +----+-----+---------------+------+------+----------+ | id | grp | ip_addr | mask | port | tag | +----+-----+---------------+------+------+----------+ | 20 | 1 | 123.17.0.0 | 23 | 5061 | Testing | | 22 | 1 | 123.17.0.0 | 23 | 5062 | Testing | | 23 | 1 | 123.17.0.0 | 23 | 5063 | Testing | | 24 | 1 | 123.17.0.0 | 23 | 5064 | Testing | | 1 | 1 | 123.17.0.0 | 23 | 5060 | | | 4 | 1 | 20.5.176.0 | 24 | 5060 | | | 21 | 1 | 123.17.1.235 | 32 | 5060 | Override | | 46 | 2 | 123.17.29.201 | 32 | 5060 | customer | +----+-----+---------------+------+------+----------+ Can anyone explain this? I¹m wondering if there¹s a bug in how allow_source_address does its net mask calculations. -H

1 0

Pseudo-Variables Digest
by Kalala Alexander 17 Dec '14

17 Dec '14

2 2

What is raw 0.0.0.0:255 pn netstat??
by Luis Jimenez 17 Dec '14

17 Dec '14

Hello, i just installed Kamailio 4.2.1 on debian Wheezy, would like to know why when Kamailio starts i see this on netstat -an ?? raw 0 0 0.0.0.0:255 0.0.0.0:* 7 is it normal?? why? Thanks in advance

3 3

How to Fix the RTP Engine SDP Rewriting direction.
by Mahmoud Ramadan Ali 16 Dec '14

16 Dec '14

Hi Dears, I'm working on integrating the rtpengine to work with Kamailio as RTP proxy and i have successfully configured the rtpengine in multi home mode to proxy the media and rewrite the SDP message whenever it passes trough the Kamailio internal interface to the external interface and vice versa using this command. root@debian:/usr/local/etc/kamailio# rtpengine --interface=Kamailio-Internal/192.168.100.1 --interface=Kamailio-External/ 192.168.50.1 --listen-ng=127.0.0.1:22222 --pidfile=/var/run/ngcp-rtpengine-daemon.pid Also i have configured the "i" and "e" flags in the route[NATMANAGE] to take care of the SDP rewriting direction the same way it worked before with my rtpproxy module configuration. But the issue now is that whenever i make a call from the external to the internal the rtpengine can NOT detect the right direction as colored in red below so my question now is : How to set the direction of the rtpengine to designate that the interface 192.168.50.1 is the external and 192.168.100.1 is the internal ? Dec 16 09:28:09 debian rtpengine[4714]: Got valid command from 127.0.0.1:50639: offer - { "sdp": "v=0#015#012o=- 13063213690566395 1 IN IP4 192.168.50.2#015#012s=X-Lite release 4.7.1 stamp 74247#015#012c=IN IP4 192.168.50.2#015#012t=0 0#015#012m=audio 56078 RTP/AVP 125 100 0 9 8 101#015#012a=rtpmap:125 opus/48000/2#015#012a=fmtp:125 useinbandfec=1#015#012a=rtpmap:100 speex/16000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012", "direction": [ "internal", "external" ], "flags": [ "asymmetric", "trust-address", "symmetric" ], "replace": [ "session-connection", "orig ... Best Regards and Thanks in advance.

2 1

IPv6 socket issues
by Kees Meijs - SIGNET B.V. 16 Dec '14

16 Dec '14

Good evening, During pre-production testing, we discovered quite odd IPv6 socket issues. Problems arise when a SIP dialog is initiated from within Kamailio (e.g. a SIP INVITE message). Using ngrep, it seems very odd IPv6 sockets are opened, for example: > U 2001:beef:34a:10::234:5060 -> 2001:beef:285:1::3:5060 1573541542@0:1440 > U 2001:beef:34a:10::234 -> 2001:beef:285:1::3 +1573541542@1448:17 Obviously the "1573541542@0:1440" part is bogus, I expect output similar to: > U 2001:beef:34a:10::234:5060 -> 2001:beef:285:1::3:5060 > INVITE sip:s@[2001:beef:285:1::3]:5060 SIP/2.0. When I do a REGISTER from UA using IPv6, the dialog (and return packets) seem completely normal. Using IPv4, there's no problem at all. It seems the socket problems are triggered when: 1. Using the topoh module. 2. When altering From: headers e.g. using uac_replace_from(). When topoh is disabled and no headers are altered, IPv6 works as good as IPv4. We could reproduce the problem using Kamailio 4.1.4, 4.1.5 and 4.2.0 on Debian (64-bit). Please advice what to do. Thanks in advance! Best regards, Kees

2 4

Re: [SR-Users] sr-users Digest, Vol 115, Issue 15
by Kees Meijs - SIGNET B.V. 16 Dec '14

16 Dec '14

Hi Daniel, Thank you very much for your prompt response. The output mentioned is generated by ngrep and can be seen by tcpdump as well. It seems a "space" and the bogus part ends up after the port used in the socket opened. For example "5060" becomes "5060 yadieyaboguspartdrdfg5r3245234". Obviously sending a packet towards an IPv6 address using such a port doesn't work. For the time being, we disabled the topoh.so module and rewrote code using uac_replace_from() and substituded the code with $fn = "string", $fd = "otherstring" equivalents. Thanks again! Cheers, Kees

1 0

Kamailio AVPs Radius
by Kalala Alexander 16 Dec '14

16 Dec '14

Radius server sends a response containing an attribute (Session-Timeout (27): 2342). How is the value saved in the AVPs using module auch_radius? Config: modparam("auth_radius", "radius_config","/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type",15) modparam("auth_radius", "use_ruri_flag", 22) modparam("auth_radius", "auth_extra", "Calling-Station-Id=$fU;Called-Station-Id=$tU;Acct-Session-Id=$ci") if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!radius_www_authorize("$td")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } ............................... if (from_uri==myself) { if (!radius_proxy_authorize("$fd", "$fU")) { proxy_challenge("$fd", "0"); exit; } -- С уважением Инженер по телекоммуникациям Калала Александр vel: 375291146285 life: 375256819996 skype: klistrod VoIP Network Engineer dCAP, CCNAVoice Kalala Alexander http://callcenters.by/ http://skytel.by/ http://voiplab.by/ http://asterisk-pbx.by/

3 10

New reminder: about private emails
by Daniel-Constantin Mierla 16 Dec '14

16 Dec '14

Given that I noticed still a lot of emails sent directly after a discussion started on public mailing lists (subject with mailing lists prefix, but the sender header is not the mailing list daemon), I would like to remind that, at least in my case, writing emails directly without cc-ing the mailing list is highly not recommended. The rule is also suggested in the mailing lists presentation page: - http://www.kamailio.org/w/mailing-lists/ Unless I asked explicitly for a private email with some specific details, the message will not be noticed in decent time and therefore not replied. I use this email address for many public mailing lists and it is intended only for that usage, I am reacting based on the filters I created, unmatched messages getting the lowest priority. Therefore if you wrote me and expect a response, re-send the message to the appropriate public mailing list. Checking and answering the messages on Kamailio mailing lists are among the top of priorities on my daily activity, so it is high chance you get the response faster when you write there. Note that you can always mask sensitive data (e.g., replace passwords, IP addresses) before sending logs or config file. There are several common sense as well as technical reasons for keeping the discussion on public mailing lists. Here are some along with other suggestions to improve the likeliness of getting an answer: - we are not answering questions on mailing lists to help only one person, but also other people that may have similar issues in the future -- they can find the answers in the archive with a search engine. Also, other people subscribed to the list can learn from conversation or contribute to the discussion, resulting in better outcome - the amount of unexpected messages is very high -- it is practically a zero chance to get to the folder with messages that didn't match any filter (thus your message is not going to be replied) - whenever I find time, the first emails I answer are those coming on mailing lists. They are many there as well, people are traveling or having other personal or business projects, so sometime is good to send a reminder if a question does not get an answer - starting with negative approach or no technical content, like you are going to use something else if you don't get this instantly, or this application is missing what so ever tutorial giving exactly the solution you need, or simply asking for full config of complex requirements, is not helping at all, but actually making people to ignore your messages. Start with what you tried and where you got stuck. Everything you get is for free, have appropriate expectation and behavior. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Email notifications from Github commits
by Daniel-Constantin Mierla 15 Dec '14

15 Dec '14

Hello, those on sr-dev mailing list probably noticed the last commit notification - the email format can be seen at: - http://lists.sip-router.org/pipermail/sr-dev/2014-December/026513.html As discussed in the past, it tries to stay as close as possible to the old notification message, but it has some differences. I published the php script I use for generating this notification as well as synchronizing the git repository mirror on kamailio.org. It is available at: - https://github.com/miconda/gitpushub If people prefer, we can make a clone inside kamailio organization and tailor it to be more specific for kamailio repository, with more people having direct access to update it. On the other hand, any contribution to the script to make it better or more useful is welcome! Eventually it is going to be useful for various people with their other projects. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Error when configuring RTPProxy-NG.
by Mahmoud Ramadan Ali 15 Dec '14

15 Dec '14

Hi Dears, I'm following this link https://github.com/sipwise/rtpengine to configure rtpproxy-ng with Kamailio BUT i'm reciving the following errors in the Kamailio log file. Dec 14 17:58:34 debian /usr/local/sbin/kamailio[6198]: ERROR: rtpproxy-ng [rtpproxy.c:1425]: rtpp_test(): proxy responded with invalid response Dec 14 17:58:34 debian rtpengine[6160]: Failed to properly parse UDP command line '6188_1 d7:command4:pinge' from 127.0.0.1:36681, using fallback RE Dec 14 17:58:34 debian /usr/local/sbin/kamailio[6188]: ERROR: rtpproxy-ng [rtpproxy.c:1425]: rtpp_test(): proxy responded with invalid response Dec 14 17:58:34 debian rtpengine[6160]: Failed to properly parse UDP command line '6202_1 d7:command4:pinge' from 127.0.0.1:38342, using fallback RE Dec 14 17:58:34 debian /usr/local/sbin/kamailio[6202]: ERROR: rtpproxy-ng [rtpproxy.c:1425]: rtpp_test(): proxy responded with invalid response Dec 14 17:58:34 debian rtpengine[6160]: Failed to properly parse UDP command line '6200_1 d7:command4:pinge' from 127.0.0.1:33265, using fallback RE My configuration as follows: root@debian:/usr/local/etc/kamailio# rtpengine --interface=Kamailio-Internal/192.168.100.1 --interface=Kamailio-External/ 192.168.50.1 --listen-udp=127.0.0.1:12222 --pidfile=/var/run/ngcp-rtpengine-daemon.pid root@debian:/usr/local/etc/kamailio# kamctl fifo nh_enable_rtpp udp: 127.0.0.1:12222 1 root@debian:/usr/local/etc/kamailio# kamctl fifo nh_show_rtpp udp:localhost:12222:: set=0 index:: 0 disabled:: 1 weight:: 1 recheck_ticks:: 0 loadmodule "rtpproxy-ng.so" modparam("rtpproxy-ng", "rtpproxy_sock", "udp:127.0.0.1:12222") modparam("rtpproxy-ng", "rtpproxy_disable_tout", 20) modparam("rtpproxy-ng", "rtpproxy_tout", 2) modparam("rtpproxy-ng", "rtpproxy_retr", 2) modparam("rtpproxy-ng", "extra_id_pv", "$avp(extra_id)") /etc/default/ngcp-rtpengine-daemon RUN_RTPENGINE=yes LISTEN_TCP=25060 LISTEN_UDP=12222 LISTEN_NG=22222 # ADDRESS=... # ADV_ADDRESS=... # ADDRESS_IPV6=... # ADV_ADDRESS_IPV6=... TIMEOUT=60 SILENT_TIMEOUT=3600 PIDFILE=/var/run/ngcp-rtpengine-daemon.pid FORK=yes # TOS=184 TABLE=0 Any ideas about how to solve this please ?! Thanks in advance and Best Regards

3 4

How to get account code
by Juned Khan 15 Dec '14

15 Dec '14

Hi All, I am using Kamailio for signalling purpose. In my current setup I am handling sip registration in Kamailio server and forwarding call to Freeswitch server , and registration and call forwarding to FS server is working. But at Freeswitch side I need accountcode to elstablish a call. I am using this to setup https://wiki.freeswitch.org/wiki/Kamailio_basic_setup_as_proxy_for_FreeSWIT… I am using Kamailio 4.2.1 and Freeswitch 1.4.14 on CentOS 6.5. I have also attached my Kamalio.cfg file. So how do I store and send accountcode variable to Freeswitch? Regards Juned Khan / Network Engineer +91 9974740823/ juned.khan(a)inextrix.com iNextrix Technologies Pvt Ltd. www.inextrix.com <http://htmlsig.com/www.inextrix.com> [image: Facebook] <https://www.facebook.com/junedkhan23> [image: Twitter] <https://twitter.com/juned23> [image: Google Plus] <https://plus.google.com/+JunedKhan> [image: Linkedin] <http://htmlsig.com/in.linkedin.com/in/junedk> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. E-mails are not secure and cannot be guaranteed to be error free as they can be intercepted, amended, or contain viruses. Anyone who communicates with us by e-mail is deemed to have accepted these risks. Company Name is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company.

2 1

Main GIT repository now on Github - https://github.com/kamailio/kamailio
by Daniel-Constantin Mierla 15 Dec '14

15 Dec '14

Hello, the main GIT repository for Kamailio project is now hosted on Github - the web link to it is: * https://github.com/kamailio/kamailio For users (read only access -- ie, you cloned from git.sip-router.org) should be no change, just be sure that DNS has been updated, because git.sip-router.org is pointing now to another server which has a real-time clone of Github repository. To know if DNS was updated on your side, browse to: * http://git.sip-router.org an you must be directed to Gitlist web interface: * http://git.kamailio.org/gitlist/index.php If you end up in an url with gitweb on it, DNS is pointing to the old server. The listed kamailio and sip-router projects are the same (the second is a symlink to the first). Gitweb is also running on the mirror, so the links from the email notifications of old commits are valid: * http://git.sip-router.org/gitweb/ Gitweb can be made the main web interface for the mirror, if people like it more. For now I let Gitlist in order to be easy to detect the DNS update. Once the DNS is updated, pulling/fetching in an existing clone done from git.sip-router.org should just work. For developers, you need to change the remote URL of your local clone to point to Githup repository. Github provides several access methods (git/ssh or https) -- as an example of switching the remote URL to github repository and do the push via https, got you your local clone directory and do: git remote set-url origin https://github.com/kamailio/kamailio.git Check the change with: git remote -v You should see: origin https://github.com/kamailio/kamailio.git (push) origin https://github.com/kamailio/kamailio.git (fetch) When you try to push, it should ask you for your github username and password. For github repository access, see more details at: - https://help.github.com/articles/which-remote-url-should-i-use/ I sent invitations to some of the developers that I found quickly on github, but there are still some of them very active that I didn't spot on github. Send me the github user id to allow you write access. >From now on all commits must be pushed to github! For fetching/pulling in an existing clone, wait for a little more or check the DNS update on your side with the above guidelines - there is already a commit I pushed there, which is no longer on the old repository (which will be decommissioned soon, but still can take a bit for the DNS to propagate, so some of you might end up to it if using git.sip-router.org). New read-only clones can be done from Github as well. Hopefully the migrations is going to be smooth for everyone! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

rtpengine offer does not replace o and c line addresses
by jh＠tutpro.com 15 Dec '14

15 Dec '14

while doing some webrtc tests, i noticed that for some reason rtpengine_offer replace-session-connection replace-origin flags don't have any effect on the ip address (0.0.0.0) at o and c lines of the sdp. the debug is below. what is it that i'm missing? -- juha Dec 15 08:26:29 rautu /usr/bin/sip-proxy[12597]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1) Dec 15 08:26:29 rautu rtpengine[10473]: Got valid command from 127.0.0.1:38653: offer - { "sdp": "v=0#015#012o=Mozilla-SIPUA-34.0 24971 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-ufrag:fb2938b0#015#012a=ice-pwd:eb43995297ab91228db0dec8949497ed#015#012a=fingerprint:sha-256 D9:DD:0F:CE:C2:D5:6A:D3:9A:C7:24:00:76:24:EA:66:ED:97:9F:E3:2D:25:23:60:67:7C:54:CF:04:8A:DB:F7#015#012m=audio 9 RTP/SAVPF 109 9 0 8 101#015#012c=IN IP4 0.0.0.0#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8 ... Dec 15 08:26:29 rautu rtpengine[10473]: ... 000#015#012a=fmtp:101 0-15#015#012a=sendonly#015#012a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level#015#012a=setup:actpass#015#012a=rtcp-mux#015#012a=candidate:0 1 UDP 2122187007 192.98.102.30 58681 typ host#015#012a=candidate:2 1 UDP 2122252543 192.168.1.194 42724 typ host#015#012a=candidate:0 2 UDP 2122187006 192.98.102.30 53515 typ host#015#012a=candidate:2 2 UDP 2122252542 192.168.1.194 34867 typ host#015#012", "ICE": "force", "replace": [ "session-connection", "origin" ], "call-id": "dgp00b8p8q45f0vba38o", "via-branch": "z9hG4bK8492226", "rec ... Dec 15 08:26:29 rautu rtpengine[10473]: ... eived-from": [ "IP4", "192.98.102.30" ], "from-tag": "a4dr7vb4on", "command": "offer" } Dec 15 08:26:29 rautu rtpengine[10473]: [dgp00b8p8q45f0vba38o] Creating new call Dec 15 08:26:29 rautu rtpengine[10473]: [dgp00b8p8q45f0vba38o] Opened ports 50110..50111 for media relay Dec 15 08:26:29 rautu rtpengine[10473]: [dgp00b8p8q45f0vba38o] Opened ports 50118..50119 for media relay Dec 15 08:26:29 rautu rtpengine[10473]: [dgp00b8p8q45f0vba38o] Returning to SIP proxy: d3:sdp717:v=0#015#012o=Mozilla-SIPUA-34.0 24971 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-lite#015#012m=audio 50110 RTP/SAVPF 109 9 0 8 101#015#012c=IN IP4 0.0.0.0#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendonly#015#012a=rtcp:50111#015#012a=rtcp-mux#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:vSMhw6pr2O+DWrL+7O1chSiCrSZ5YJW4esLlgNv/#015#012a=setup:actpass#015#012a=fingerprint:sha-1 42:C4:38:DC: ... Dec 15 08:26:29 rautu rtpengine[10473]: [dgp00b8p8q45f0vba38o] ... A6:4A:A6:70:49:6A:96:81:BE:9A:17:7D:FE:02:00:2C#015#012a=ice-ufrag:GL0QSUR3#015#012a=ice-pwd:X1irWG0YUiMpcTLgHo9OHgNl1NbT#015#012a=candidate:Wz1BLNSFenkyID4v 1 UDP 2130706431 0.0.0.0 50110 typ host#015#012a=candidate:Wz1BLNSFenkyID4v 2 UDP 2130706430 0.0.0.0 50111 typ host#015#0126:result2:oke

2 1

Change Invite message to 302 Moved
by Leky Shakur 14 Dec '14

14 Dec '14

Hello, I have a situation, where provider does not understand reivnites when user makes a Call Forwarding via Asterisk server, it only understands 302 Moved messages in CF situations (or in Call tranfer). So my scenario is like this: Provider sends Invite to Kamailio, I forward it to Asterisk server. where a user make a call forwarding to Kamailio (INVITE message), who then forwards it to provider - my question if, can I change the INVITE that Asterisk send Kamailio server and change it to 302 Moved message and relay it to provider (I can detect when there is an CF, because Asterisk sends diversion header in the INVITE towards Kamailio)?

1 0

t_suspend/t_continue
by Ahmed Salem 13 Dec '14

13 Dec '14

Hi, I have 2 kamailio servers behind a load balancer. How can I use t_suspend on one server and t_continue on the other one. I already have the index and label of the transaction saved in a common database shared by the two kamailio servers. The reason I want to do this is for push notifications for iOS and Android, and I can't guarantee that the caller and the callee will register on the same server because of the load balancer. Thanks Ahmed

2 5

Error When Configuring the sipcapture module.
by Mahmoud Ramadan Ali 13 Dec '14

13 Dec '14

Hi Dears, I'm trying to configure the SIP Capture module with Kamailio BUT i'm receiving that Error. Dec 13 09:20:53 debian /usr/local/sbin/kamailio[5183]: ERROR: sipcapture [hep.c:84]: hep_msg_received(): ERROR: sipcapture:hep_msg_received: not supported version or bad length: v:[82] l:[69] Any ideas how to solve this please ?! Best Regards and Thanks in advance

1 0

kamailio for newbies!
by Aria Mill 13 Dec '14

13 Dec '14

Hello, Kamailio is difficult. Many of us want to use it because is open source and it's flexible. but to tell you the truth after 3-4 hours of playing with it I am frustrated, and I am starting to hate it! Why has nobody made a kamailio video series on youtube?-example of how to install kamailio-example of how to admistrer kamailio, and what other programs you need to work with;-example how to create users in Kamailio for enterprise use.-example of running kamailio outside enterprise like a SBC with a public ip;-example of how can you run kamailio as a load balance server.-example how to save money on your sip infrastructure using kamailio instead of cisco cube for example or sonus..or any other example you might thing is useful in a sip environment! I look at youtube and I see Kamailio this and kamailio that, but no actual tutorial! This is 2014, people don’t waste time reading documentation, youtube is much more efficient...I think Kamailio will not have the success of it's potential until senior community members will not take the plunge and broadcast their knowledge on a proper video channel! Asterisk has every scenario up there based on these info I was able to setup in less that 3 days my office and 40 people are making calls......It's unbelievable that after so many years nobody has made a godam explicit tutorial for newbies in kamailio...I am so frustrated right now!!

8 12

Create and control a new dialog to different server during main dialog
by Konstantin M. 12 Dec '14

12 Dec '14

Hello, I want to achieve the following schema: 1. Accept an initial invite 2. Create separate dialog to different SIP server (probably with some headers from point 1) 3. Wait until that different SIP server will complete dialog. 4. Based on result of point 3, transfer invite from 1 to the next steps or decline a call. The problem I've experienced - how to create a separate dialog and control it inside main dialog? Thanks a lot!

2 2

Planning to switch committing to github
by Daniel-Constantin Mierla 11 Dec '14

11 Dec '14

Hello, I setup the mirror of Kamailio github repository to kamailio.org, can be seen online at: - http://git.kamailio.org/gitlist/index.php/kamailio/ Mirroring is triggered by a push to github, so it is pretty much in real time. Right now the commit and mirroring flow is: - commit to git.sip-router.org triggers mirroring (push) to github, which triggers a mirroring (fetch) on git.kamailio.org Practically with a commit pushed directly to github, the first part of the above is skipped. All seems to be ok, my last commits were done to test the mirroring including the notification emails - I will send a separate message about what we can get there. The missing parts are the deployment of gitweb in order to have the links from archived email notifications still valid and pointing git.sip-router.org to kamailio mirror server. Considering all above, I propose to stop committing to git.sip-router.org and pushing to github starting either next Monday (Dec 15) (or Wednesday (Dec 17) if not ready by Monday). Existing developers have to send me the github username to add it to kamailio project. Any suggestions and comments are welcome! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 4

Charging System
by Abdul Gafar 11 Dec '14

11 Dec '14

Hi all I'm trying install IMS charging module,is this posible use kamailio server, without install other module IMS ? //Gafar

2 2

Appending/removing a value associated with domain
by Olli Heiskanen 11 Dec '14

11 Dec '14

Hello, Something I've been wondering about meddling with sip uris with Kamailio: I know it's possible to translate between a number prefix and a domain using PDT, but is this possible: Having a numeric or alphanumeric value stored in db, associated to a domain and appended to / removed from a number during processing a sip message? For example 112222333(a)mydomain.com --> 333.5555(a)mydomain.com where value 5555 and mydomain.com are associated to one another in db somewhere. In this example I could have a pdt with prefix 112222 and domain mydomain.com, but the value 5555 should be stored somewhere as well. So one part is to find a db table suitable for storing a value and associating it with a domain, and the other part is to change the sip uris. I'll appreciate any comments! cheers, Olli

1 0

Federation Kamailio
by nadie 11 Dec '14

11 Dec '14

Hello. I installed kamailio. All OK. But when i call to other servers. ostel.co. for example. no ring. What element permits federation between kamailios? DNS Records? Thank you.

3 3

Kamailio - change SIP type
by Leky Shakur 11 Dec '14

11 Dec '14

Hello, Is is possible to change SIP message type? Example: Server A sends Kamailio SIP INVITE message, can I change this SIP INVITE to "302 - MOVED" (and reuse data from SIP INVITE) and send it to Server B? Does anybody have any example? Best Regards.

1 0

Kamailio Truncating timestamp of sip message?
by Heenan, Timothy Steven 10 Dec '14

10 Dec '14

Greetings, I've recently run into a rather strange issue where, in response to an INVITE sent by Kamailio, I'm receiving a 183 Session Progress and SIP 200 OK response about .000050 seconds apart (.05 miliseconds) . Kamailio is choosing to forward the SIP 200 OK ahead of the 183, as if they had arrived at the same time and is prioritizing the 200. Is there a truncate limit on timestamp values? Is this possibly adjustable? Thank you in advance, -Tim ---------------------------------------------------------------------- This email message and any attachments are for the sole use of the intended recipient(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message and any attachments.

2 1

RFC: infrastructure upgrade - git, tracker, ...
by Daniel-Constantin Mierla 10 Dec '14

10 Dec '14

Hello, as most of you know, we have a rather distributed infrastructure, with servers provided from different companies or persons. We came to the time when one of the servers is too old and considered to be decommisioned, so we have to decide how to move on. It is about sip-router.org, who was offered/sponsored by courtesy of Jan Janak. From project point of view, the server hosts: - git repository - bug tracker - website and wiki under domain sip-router.org Given that no matter what we like, there is work to do, I am looking to see what are the best options out there for everyone involved in the project. The sip-router.or wbesite and wiki, which are not really updated anyhow, will be relocated as virtual host in kamailio.org server and made static for historic purposes. For git and tracker, I thought of two variants: 1) move to use an external hosting service - the first candidate and perhaps the only to be considered is GitHub, we have there already a real time mirror of git repository. Then we should get a read only mirror to kamailio.org. If the tracker on github is good enough for everyone, then we will use it. I could see quickly that lot of kamailio developers already have an account on github. 2) get a new server and relocate those components there. It will need to be configured from scratch and the components eventually updated to use latest versions. In case of tracker, we have eventually to re-evaluate if flyspray worth keeping, as we had several discussions, due to the fact that the project doesn't seem to be very active. My personal preference at this moment is 1), given that offloads administration works from project. For 2) we will need someone to commit to a sysadmin job for long time. As probably you noticed lately, serious security vulnerabilities can appear and someone needs to take care of proper maintenance of the server. I don't want to get all services on kamailio.org, as it has other critical components (mailing lists, releases, website, ...) and messing it or overloading doesn't make sense. There is no real pressure to come to a decision, we can still rely on the server for a while, but I would rather not postpone it for long. While users are encouraged to give their opinion, I feel that existing developers should have the main role in decision, being something that impacts them directly. Your preference? Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com

8 12

Kamailio Crash when modifying username of Request Uri
by Timo Klecker 10 Dec '14

10 Dec '14

Hi all! Our Kamailio (3.1.4) crashed when modifying the username-Part of a Request URI using: $rU = $(rU{s.strip,6}); Initial Request URI: /sbin/kamailio[40658]: NOTICE: <script>: [ INVITE ] Got new Packet to sip:123456:pass@sip.custom.com The config-script now removed the first 6 chars from the username part and would replace them in one of the following lines, but Kamailio had problems here: /sbin/kamailio[40658]: ERROR: pv [pv_core.c:251]: failed to parse the R-URI /sbin/kamailio[40658]: NOTICE: <script>: Strip from Head: Request Username: <null> // $rU /sbin/kamailio[40658]: ERROR: pv [pv_core.c:251]: failed to parse the R-URI /sbin/kamailio[40658]: ERROR: <core> [action.c:801]: ERROR: do_action: bad uri <sip::passsip.custom.com>, dropping packet Is there maybe a error when modifying a uri with password this way? Do we have to change this? Ultimately Kamailio crashed: /sbin/kamailio[40658]: ERROR: <core> [action.c:1513]: run action error at: :0 /sbin/kamailio[40658]: ERROR: pv [pv_core.c:1761]: do action failed /sbin/kamailio[40658]: ERROR: <core> [lvalue.c:358]: setting pvar failed /sbin/kamailio[40658]: ERROR: <core> [lvalue.c:411]: assignment failed at pos: (319,10-319,34) /sbin/kamailio[40658]: ERROR: pv [pv_core.c:251]: failed to parse the R-URI Like 5000 lines more of /sbin/kamailio[40658]: ERROR: pv [pv_core.c:251]: failed to parse the R-URI Due to rerouting internal and trying to remove more parts of the request Uri using $rU = $(rU{s.strip,6}); Which then concluded to the following: /sbin/kamailio[40656]: ALERT: <core> [main.c:741]: child process 40660 exited by a signal 11 /sbin/kamailio[40656]: ALERT: <core> [main.c:744]: core was generated /sbin/kamailio[40656]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD /sbin/kamailio[40659]: ERROR: pv [pv_core.c:251]: failed to parse the R-URI /sbin/kamailio[40682]: INFO: <core> [main.c:807]: INFO: signal 15 received /sbin/kamailio[40677]: INFO: <core> [main.c:807]: INFO: signal 15 received /sbin/kamailio[40676]: INFO: <core> [main.c:807]: INFO: signal 15 received /sbin/kamailio[40678]: INFO: <core> [main.c:807]: INFO: signal 15 received [...] I will try to recreate this using the new version (4.1.6) on monday. Kind regards Timo Klecker

3 23

Sanity Module - no 400 Bad Request message when no VIA header
by Kowalski Piotr 3 - Hurt 10 Dec '14

10 Dec '14

Hello everybody, We are facing a problem with Sanity Module. We need to use it to force Kamailio to send back 400 Bad Request message when there's no VIA header within INVITE. How can we achieve this? In our opinion there's no possibility to set this easy. We get no answer from Kamailio when sending corrupted INVITE (with no VIA headers). Many thanks for your help. Piotr Kowalski Starszy Architekt Techniczny Tel. +48 22 699 50 93 Kom. +48 519 123 289 Orange Orange Polska Hurt Badania i Rozwój Sieci i Platform Usługowych Wydział Usług Konwergentnych Obrzeżna 7 /p. 309, 02-691 Warszawa Fax. +48 22 857 99 86 www.hurt-orange.pl<http://www.hurt-orange.pl/> - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Czy musisz drukować tę wiadomość? Pomyśl o środowisku. Korzystaj z e-faktury. Treść tej wiadomości jest własnością Orange Polska i zawiera informacje stanowiące tajemnicę przedsiębiorstwa Orange Polska. Jeżeli nie jesteście Państwo jej adresatem, bądź otrzymaliście ją przez pomyłkę, prosimy o powiadomienie o tym nadawcy oraz trwałe jej usunięcie. Orange Polska Spółka Akcyjna z siedzibą i adresem w Warszawie (02-326) przy Al. Jerozolimskich 160, wpisana do Rejestru Przedsiębiorców prowadzonego przez Sąd Rejonowy dla m.st. Warszawy XII Wydział Gospodarczy Krajowego Rejestru Sądowego pod numerem 0000010681; REGON 012100784, NIP 526-02-50-995; z pokrytym w całości kapitałem zakładowym wynoszącym 3.937.072.437 złotych.

2 1

Registration at SCSCF Failed
by Kamal Palei 10 Dec '14

10 Dec '14

Hi All In 4.2.0 Kamailio (S-CSCF), when I try to register an IMS endpoint, I get 403 from P-CSCSF. When I checked the error logs, SCSCF logs look as below 1(5455) INFO: ims_auth [authorize.c:824]: authenticate(): uri=sip: group4-imslab11.in nonce=b9eec0ad9d6d186408541d8a23a7c7a5 response=df89adebbf28f25d38d2f993f7cde033 qop= nc= cnonce= hbody= 1(5455) INFO: ims_auth [authorize.c:872]: authenticate(): UE said: df89adebbf28f25d38d2f993f7cde033 and we expect df89adebbf28f25d38d2f993f7cde033 ha1 599fb074cf189867eed06f3b6ebc0b97 (REGISTER) 1(5455) INFO: ims_registrar_scscf [cxdx_sar.c:79]: create_return_code(): created AVP successfully : [saa_return_code] - [-2] 12(5466) INFO: ims_registrar_scscf [cxdx_avp.c:138]: cxdx_get_avp(): cxdx_get_experimental_result_code: Failed finding avp 12(5466) INFO: ims_registrar_scscf [cxdx_avp.c:138]: cxdx_get_avp(): cxdx_get_charging_info: Failed finding avp I have added an user necs_0 in hss_db, so why I get below error. Please let me know what is missed here. Is S-CSCF not able to contact HSS ?? Best Regards kamal

2 5

Kamailio Crash
by Kristian Kielhofner 09 Dec '14

09 Dec '14

version: kamailio 4.1.6 (x86_64/linux) 58746c I looked through 4.1 commits but didn't catch anything that looked like a fix for this but it's a very real possibility this is a duplicate (again). Any thoughts (backtrace attached)? Thanks! -- Kristian Kielhofner

2 6

Contact Header User
by Marc Soda 09 Dec '14

09 Dec '14

Can someone recommend a way to extract the user part of a Contact header URI? Right now I'm just trying to pass it to xlog(). I have tried: - xlog("L_NOTICE", "$(ct{uri.user})"); This gives me an empty string. - xlog("L_NOTICE", @contact.uri.user); This gives me an error starting Kamailio: "function xlog: parameter 2 is not constant". Note, I can't seem to get any select working, for example, @ruri gives me the same error. Any suggestions? Thanks, Marc

3 3

[Kamailio -feature]
by Ludovic DURAND 09 Dec '14

09 Dec '14

Hi, The purpose of projet is to implement a VoIP secure solution with Kamailio as core IMS network. In this project VoIP call will be established with zRTP. SBC is transparent for zRTP call, zRTP is establish between 2 end point (terminals), SBC don't decrypt or encrypt the call. Thus SBC don't modify codec. Is there a SBC solution provided by Kamailio ? How is managed the SBC failover mecanism ? is there SBC backup ? What is the status of context registered in Kamailio (P-CSCF & S-CSCF) in case of SBC failure ? - Is it deleted automatically ? - dispatched to another SBC (backup SBC) ? Is there mecanism in Kamailio to check SBC overload and manage a dispatching of zRTP call to available SBC ? Question related to P-CSCF and S-CSCF : - Is there redundant function (such as active/active or active/standby solution) of contextual data (customer data stored at the registration REGISTER on P-CSCF and S-CSCF) ? Best Regards Ludovic DURAND SII

2 1

acc.so segfault using 4.3.0-dev2 (6daafdb5b6c9fc72698d43f81656bd511cacce69)
by Anthony Messina 08 Dec '14

08 Dec '14

I get repeatable segfaults with acc.so in 4.3.0-dev2. Fedora 20 x86_64. Debug log attached. -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

2 7

Kamailio 4.2 crash
by Rahul MathuR 08 Dec '14

08 Dec '14

Hello, I have recently moved some traffic to kamailio-4.2 (installed on a seperate box) and it crashes quite often. Below is the backtrace of the core files - <CORE#1> (gdb) bt #0 0x00000033d8c32635 in raise () from /lib64/libc.so.6 #1 0x00000033d8c33e15 in abort () from /lib64/libc.so.6 #2 0x000000000049add5 in sig_alarm_abort (signo=14) at main.c:689 #3 <signal handler called> #4 0x00000033d8ce50c7 in syscall () from /lib64/libc.so.6 #5 0x00007f7ca2ea0a1e in futex_get (lock=0x7f7ca3334388) at ../../mem/../futexlock.h:110 #6 0x00007f7ca2ea1e99 in remove_dialog_timer (tl=0x7f7ca3384c98) at dlg_timer.c:168 #7 0x00007f7ca2e83e9b in destroy_dlg (dlg=0x7f7ca3384c40) at dlg_hash.c:357 #8 0x00007f7ca2e84e19 in destroy_dlg_table () at dlg_hash.c:438 #9 0x00007f7ca2e50b89 in mod_destroy () at dialog.c:773 #10 0x000000000058dfd5 in destroy_modules () at sr_module.c:818 #11 0x0000000000499ad3 in cleanup (show_status=1) at main.c:564 #12 0x000000000049b096 in shutdown_children (sig=15, show_status=1) at main.c:706 #13 0x000000000049d66c in handle_sigs () at main.c:797 #14 0x00000000004a4e57 in main_loop () at main.c:1757 #15 0x00000000004a8a43 in main (argc=2, argv=0x7fff58d21368) at main.c:2609 (gdb) quit <CORE#2> (gdb) bt #0 0x00007f7ca2ea22d7 in update_dlg_timer (tl=0x58, timeout=10) at dlg_timer.c:203 #1 0x00007f7ca2e824f3 in dlg_clean_run (ti=44865910) at dlg_hash.c:253 #2 0x00007f7ca2e566fc in dlg_clean_timer_exec (ticks=44865910, param=0x0) at dialog.c:1250 #3 0x00000000005f93e8 in fork_sync_timer (child_id=-1, desc=0x7f7ca2eb3a71 "Dialog Clean Timer", make_sock=1, f=0x7f7ca2e566e3 <dlg_clean_timer_exec>, param=0x0, interval=90) at timer_proc.c:232 #4 0x00007f7ca2e50476 in child_init (rank=0) at dialog.c:730 #5 0x000000000058e385 in init_mod_child (m=0x7f7ca8c09eb8, rank=0) at sr_module.c:930 #6 0x000000000058e0c0 in init_mod_child (m=0x7f7ca8c0b000, rank=0) at sr_module.c:927 #7 0x000000000058e68f in init_child (rank=0) at sr_module.c:957 #8 0x00000000004a435c in main_loop () at main.c:1705 #9 0x00000000004a8a43 in main (argc=2, argv=0x7fff58d21368) at main.c:2609 (gdb) quit kamailio -v version: kamailio 4.2.0-pre0 (x86_64/linux) a78764-dirty flags: STATS: Off, USE_TCP, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_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. id: a78764 -dirty Kindly let me know which version to move. -- Warm Regds. MathuRahul -- Warm Regds. MathuRahul

2 3

Git commit notifications
by Daniel-Constantin Mierla 08 Dec '14

08 Dec '14

Hello, in the perspective of moving main git repository to github (where the commits will be done), with a real time mirror on kamailio.org, I am pasting here the commit email notifications that are now available, to decide on which one to use in the future -- see them at the end of the email. A) - this is what we get now from git.sip-router.org B) - this is what github sends C) - this is sent by kamailio.org when it gets the trigger (on a push to gihub) to mirror. It is done by a php script I wrote over the weekend, which does also the mirroring. The script that can be improved, I am going to push it on github these days, after a bit of polishing, so everyone can contribute to it. Right now it tries to be as close as possible to A), with: - first part of commit id in subject (github does it, thought it can be handy sometime) - links to raw diff and patch - it is missing the commit changes summary/statistics (probably can be added in the future with a bit of php coding) Given the flexibility and that it gives more details, pretty close to A), I will go to C). But if there are people simply liking B), then we can use it, being done by github, it's no overhead for us. Other opinions? Cheers, Daniel A) Current commit email notification: Subject: git:master: auth_db: added sections ids for functions Body: Module: sip-router Branch: master Commit: d17f162ee565b80ecd46b6f8f916a683748bf08b URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=d17f162… Author: Daniel-Constantin Mierla <miconda(a)gmail.com> Committer: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Mon Dec 8 09:11:32 2014 +0100 auth_db: added sections ids for functions - better reference for web indexing --- modules/auth_db/doc/auth_db_admin.xml | 13 ++++++------- 1 files changed, 6 insertions(+), 7 deletions(-) diff --git a/modules/auth_db/doc/auth_db_admin.xml b/modules/auth_db/doc/auth_db_admin.xml index a6f667e..74d6d00 100644 --- a/modules/auth_db/doc/auth_db_admin.xml +++ b/modules/auth_db/doc/auth_db_admin.xml @@ -304,7 +304,7 @@ modparam("auth_db", "version_table", 0) <section> <title>Functions</title> - <section> + <section id="auth_db.f.www_authenticate"> <title> <function moreinfo="none">www_authenticate(realm, table [, method])</function> </title> @@ -395,7 +395,7 @@ if (!www_authorize("kamailio.org", "subscriber")) { </example> </section> - <section> + <section id="auth_db.f.www_authorize"> <title> <function moreinfo="none">www_authorize(realm, table)</function> </title> @@ -406,7 +406,7 @@ if (!www_authorize("kamailio.org", "subscriber")) { </para> </section> - <section> + <section id="auth_db.f.proxy_authenticate"> <title> <function moreinfo="none">proxy_authenticate(realm, table)</function> </title> @@ -469,7 +469,7 @@ if (!proxy_authorize("$fd", "subscriber)) { </example> </section> - <section> + <section id="auth_db.f.proxy_authorize"> <title> <function moreinfo="none">proxy_authorize(realm, table)</function> </title> @@ -480,7 +480,7 @@ if (!proxy_authorize("$fd", "subscriber)) { </para> </section> - <section> + <section id="auth_db.f.auth_check"> <title> <function moreinfo="none">auth_check(realm, table, flags)</function> </title> @@ -557,9 +557,8 @@ if (!auth_check("$fd", "subscriber", "1")) { </example> </section> - <section> - <title> + <title id="auth_db.f.is_subscriber"> <function moreinfo="none">is_subscriber(uri, dbtable, flags)</function> </title> <para>The function checks if there is a subscriber corresponding to B) Github commit email notification: Subject: [kamailio/kamailio] d17f16: auth_db: added sections ids for functions Body: Branch: refs/heads/master Home: https://github.com/kamailio/kamailio Commit: d17f162ee565b80ecd46b6f8f916a683748bf08b https://github.com/kamailio/kamailio/commit/d17f162ee565b80ecd46b6f8f916a68… Author: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: 2014-12-08 (Mon, 08 Dec 2014) Changed paths: M modules/auth_db/doc/auth_db_admin.xml Log Message: ----------- auth_db: added sections ids for functions - better reference for web indexing C) Custom commit email notification from kamailio.org Subject: git:master:d17f162e: auth_db: added sections ids for functions Body: Module: kamailio Branch: master Commit: d17f162ee565b80ecd46b6f8f916a683748bf08b URL: https://github.com/kamailio/kamailio/commit/d17f162ee565b80ecd46b6f8f916a68… Author: Daniel-Constantin Mierla <miconda(a)gmail.com> Committer: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: 2014-12-08T09:11:32+01:00 auth_db: added sections ids for functions - better reference for web indexing --- Modified: modules/auth_db/doc/auth_db_admin.xml --- Diff: https://github.com/kamailio/kamailio/commit/d17f162ee565b80ecd46b6f8f916a68… Patch: https://github.com/kamailio/kamailio/commit/d17f162ee565b80ecd46b6f8f916a68… --- diff --git a/modules/auth_db/doc/auth_db_admin.xml b/modules/auth_db/doc/auth_db_admin.xml index a6f667e..74d6d00 100644 --- a/modules/auth_db/doc/auth_db_admin.xml +++ b/modules/auth_db/doc/auth_db_admin.xml @@ -304,7 +304,7 @@ modparam("auth_db", "version_table", 0) <section> <title>Functions</title> - <section> + <section id="auth_db.f.www_authenticate"> <title> <function moreinfo="none">www_authenticate(realm, table [, method])</function> </title> @@ -395,7 +395,7 @@ if (!www_authorize("kamailio.org", "subscriber")) { </example> </section> - <section> + <section id="auth_db.f.www_authorize"> <title> <function moreinfo="none">www_authorize(realm, table)</function> </title> @@ -406,7 +406,7 @@ if (!www_authorize("kamailio.org", "subscriber")) { </para> </section> - <section> + <section id="auth_db.f.proxy_authenticate"> <title> <function moreinfo="none">proxy_authenticate(realm, table)</function> </title> @@ -469,7 +469,7 @@ if (!proxy_authorize("$fd", "subscriber)) { </example> </section> - <section> + <section id="auth_db.f.proxy_authorize"> <title> <function moreinfo="none">proxy_authorize(realm, table)</function> </title> @@ -480,7 +480,7 @@ if (!proxy_authorize("$fd", "subscriber)) { </para> </section> - <section> + <section id="auth_db.f.auth_check"> <title> <function moreinfo="none">auth_check(realm, table, flags)</function> </title> @@ -557,9 +557,8 @@ if (!auth_check("$fd", "subscriber", "1")) { </example> </section> - <section> - <title> + <title id="auth_db.f.is_subscriber"> <function moreinfo="none">is_subscriber(uri, dbtable, flags)</function> </title> <para>The function checks if there is a subscriber corresponding to -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

rtpengine use of udp
by Juha Heinanen 05 Dec '14

05 Dec '14

what has been the motivation that rtpengine control protocol is using udp instead of tcp or ssl? we have noticed that reply to delete containing stats is too large for a single udp packet over ethernet and causes udp fragmentation and fragmentation needed requests. -- juha

2 1

OpenBiz error signing into Siremis 4.1
by Daniel Shoemaker 04 Dec '14

04 Dec '14

When I try to sign into Siremis 4.1 using admin and the default password it gives me a pop-up saying that Openbiz has encountered an error: [{"target":"ERROR","content":" \n[2014-11-17 11:51:15 (GMT)] An exception occurred while executing this script: \nError message: #8192, Non-static method BizSystem::dbConnection() should not be called statically, assuming $this from incompatible context<\/font> \nScript name and line number of error: \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php:423<\/font> \n *function:<\/b> errorHandler ( 8192, \"Non-static method BizSystem::dbConnection() should not be called...\", \"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\", 423, Array(0) ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/sysheader.inc 117\nfunction:<\/b> userErrorHandler ( 8192, \"Non-static method BizSystem::dbConnection() should not be called...\", \"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\", 423, Array(0) ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php 423\nfunction:<\/b> getDBConnection ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Lite.php 406\nfunction:<\/b> _run_search ( Array(2) ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Lite.php 345\nfunction:<\/b> fetchRecords ( \"[username]='admin' and status='1'\", Array(0), 1 ) @ \/var\/www\/siremis-4.1.0\/siremis\/modules\/service\/authService.php 98\nfunction:<\/b> authDBUser ( \"admin\", \"admin\" ) @ \/var\/www\/siremis-4.1.0\/siremis\/modules\/service\/authService.php 66\nfunction:<\/b> authenticateUser ( \"admin\", \"admin\" ) @ \/var\/www\/siremis-4.1.0\/siremis\/modules\/user\/form\/LoginForm.php 92\nfunction:<\/b> Login ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php 310\nfunction:<\/b> invoke ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php 110\nfunction:<\/b> dispatchRequest ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php 32\nfunction:<\/b> include_once ( \"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php\" ) @ \/var\/www\/siremis-4.1.0\/siremis\/bin\/controller.php 6<\/div>\n------------------------------Please ask system administrator for help...<\/div>\n"}]* My web server is configured to use apache and I've checked the ownership on the files and folder. I did a chmod 775 starting at the siremis-4.1.0/ directory. I did a search and found that the openbiz folder needs to be right under the siremis-4.1.0/ folder which it is. I have tried reinstalling only to get the same result. I'm currently running CentOS 7, Kernel 3.10.0-123.9.3, php 5.6.2-3, mariadb 10.0.14-1, & httpd 2.4.6-18. Dan -- Daniel Shoemaker 501 Claire Street Hagerstown, MD 21740 (240) 527-7054 dancoe77(a)gmail.com http://www.google.com/

5 7

Registrar path support on lookup
by Ben Langfeld 04 Dec '14

04 Dec '14

I'm facing a problem which I believe may be a bug in the registrar module's lookup. When a registration comes with multiple Path headers, these are recorded in the location table as comma separated. When the lookup is performed and these are used to construct the Route header(s) on an INVITE, this comma-separated list is not split into multiple Route headers, but instead included verbatim in a single header. Here comes my theory for how this breaks my scenario: When the INVITE then reaches the first hop (first Path header, also Kamailio), relay() sees that the Route header is itself and/or sees that there is only one Route header. The observed result is that the first hop then sends the INVITE directly to the Contact, instead of via the second hop. An example of the situation I'm facing is shown in https://gist.github.com/benlangfeld/b374a0ce0bdb6bdc35e7. Is there a particular reason the combined path is not split on lookup? Is this a bug / oversight? Am I crazy? Thanks! Ben

3 4

Re: [SR-Users] Not getting the final response after ringing
by Rizwan Khan 04 Dec '14

04 Dec '14

I am not sure if its the issue on my client end as CSIPSImple is not sendin the PRACK that is expected by the server or the rtp ports are not opened. I guess the ports are opened as I can hear two way voice. But the following issues remain: 1. I cannot receive a call on the SIP number 2. I dont get an indication on the caller (thru CSIPSimple) that a call has started as in it doesnt start the call timer 3. The call disconnects with an error after some time ..INternal Server Error. Regards, Rizwan Khan On Mon, Dec 1, 2014 at 11:00 AM, Rizwan Khan <rizkhan(a)gmail.com> wrote: > Thanks for the help guys. > > I used a stateless proxy and was able to register. Now I am able to make > calls as well but there is an issue. > > I receive multiple ringing but not receiving 200 ok after that. I keep on > receiving the ringing messages. Although the call gets connected and I hear > two way voice as well. > > Also I am call through the sip number, but its not receiving any calls. > > Any ideas? > > Rizwan Khan > > > > On Thu, Nov 27, 2014 at 9:04 PM, Daniel Tryba <daniel(a)pocos.nl> wrote: > >> On Thursday 27 November 2014 08:29:51 Rizwan Khan wrote: >> > We want to use Kamailio as the pass-thru proxy in front of the SBC just >> to >> > receive the request on a random port, and then forward the request to >> the >> > SBC internally being on the same private network. >> > >> > What would be the best way to do it? >> >> If it is the best is up to the experts, but a way to do is to check >> wheter the >> source is the SBC or not. If the message is from the SBC just handle it >> statefully (t_relay), if it isn't from the SBC set the requested domain >> to the >> ipaddr of the SBC (10.1.2.3) and process stateful. You could use the >> received >> port instead if external clients only communicate on the special port >> ($Rp!=6334) >> >> kamailio.cfg: >> >> #... >> port=5060 >> port=6334 >> #... >> >> request_route { >> #... >> >> if($si!="10.1.2.3") >> { >> $rd="10.1.2.3"; >> } >> >> route(RELAY); >> } >> >> >> >> _______________________________________________ >> 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 >> > >

2 2

Kamailio Failed to Query the Asterisk DB to Authenticate Registration Requests.
by Mahmoud Ramadan Ali 04 Dec '14

04 Dec '14

Hi Dears, I'm trying to configure Kamailio as SBC in multi home mode for Asterisk by authenticating the inbound SIP registration requests,i'm following this tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb to achieve this goal. i have modified the necessary changes like the Asterisk DB URL and the SIP table name and Username and password column and verified the Mysql connection. My topology like this Asterisk (192.168.100.10) <----Internal:192.168.100.1---->Kamailio<---External:192.168.50.1-----> SIP Phone (192.168.50.2) But when trying to register a SIP phone Kamailio generates these error messages Nov 22 17:19:20 debian /usr/local/sbin/kamailio[6434]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Unknown column 'secret' in 'field list' Nov 22 17:19:20 debian /usr/local/sbin/kamailio[6434]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Nov 22 17:19:20 debian /usr/local/sbin/kamailio[6434]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database The Auth_DB Params in the configuration file is as follows. # ----- 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", "account") modparam("auth_db", "password_column", "secret") 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 I tried also to give the Auth_DB module all the paramaters like this but still receiving the same error ! So any ideas about the cause of the problem and how to solve it please ? # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "user_column", "account") modparam("auth_db", "password_column", "secret") modparam("auth_db", "db_url", "mysql:// sipuser:sippassword@192.168.100.10/asterisk") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "account") modparam("auth_db", "password_column", "secret") modparam("auth_db", "db_url", "mysql:// sipuser:sippassword@192.168.100.10/asterisk") modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", "db_url","mysql:// sipuser:sippassword@192.168.100.10/asterisk") modparam("auth_db", "password_column", "secret") modparam("auth_db", "user_column", "account") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif

2 2

Re: [SR-Users] Evapi reply processing
by DanB 04 Dec '14

04 Dec '14

Thanks Daniel, This was exactly what I was looking for. I have also reviewed again your "Asynchronous Processing in Kamailio Configuration File" presentation from Kamailio World 2014 and it provided me an incredible amount of digested information regarding async processing(especially the part with push notification). Will follow that approach, and yes, it is a brilliant idea to pass the transaction indentifiers to CGR and process them in evapi reply route, so no more dependency on htable needed. Cheers, DanB

1 0

Refreshing the Kamailio logo
by Daniel-Constantin Mierla 04 Dec '14

04 Dec '14

Hello, I was considering to discuss the state of Kamailio project logo for a while. The current state is ambiguous, to say at least - various logos in different places. When OpenSER was renamed in Kamailio we were trying to get something new, but due to the 'forks and dark waters' at that time, this process was never completing, meanwhile using an adaptation of old openser logo. However, there are no high resolution or other variants suitable for using in particular conditions (e.g., black and white, big posters). First, a question that we should decide on: shall we aim for a new logo or try to get someone to make proper graphics from the current shape? (for reference, the logo from http://www.kamailio.org/wp-images/kamailio-rock-logo.jpg). I would prefer to go for a new logo, but I admit that this process can become complex, a mater of tastes, not something we can decide on technical aspects. Therefore, if it ends up in something that doesn't seem to have the acceptance of major contributors (with coding, advertising/promoting, ...) to the project, then we stick with current for a while. My preference is based on the fact that Kamailio evolved over the time, has its own identity, so a fresh look of the logo would reflect that better. I am listing here the options, that people can express preference on: A) New logo A1) variant based on Kamailio World logo (used by twitter kamailioproject account and on some materials at various events). It was a backup variant so far as we had high resolution graphics that could be adapted to get a good image for big posters. Can be seen at: - http://www.kamailio.org/pub/kamailio-logos/variants/kamailio-from-world-log… - http://www.fredposner.com/1471/kamailio-at-astricon-2013/ A2) use a professional service for that. Kamailio World logo was made with 99designs.com service, which is quite good considering there will be many submissions to choose between, options can be visible by everyone. There might be other similar services, this is one I have experience with and worked fine in the past -- anyhow, it has to be something that all management and solution can be done online. Asipto can sponsor at least the bronze package if 99designs is going to be used, maybe others want to participate with extra funds to get a better package. A3) leverage on community, in case some companies have internal designers or some of you have such skills or friends. Perhaps this has the highest risk on escalating, given that is going to be some direct involvement in proposals and selection. B) Old logo B1) get someone to produce high resolution graphics from existing logo No matter what variant, who will do the work or financial contribution, will get credits via our web site. However, all the rights of the resulting logo will belong to the Kamailio project. To conclude, the order of my preferences: A2, A1, A3, B1 Looking forward to feedback from many of you! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 3

Evapi reply processing
by DanB 04 Dec '14

04 Dec '14

Hey Guys, We are in the process of implementing prepaid control using CGRateS engine and for that we will be using evapi module (thanks Daniel again for that). I was wondering if you can advice us whether the path we are taking is the correct one. On INVITE, we will initiate a MaxCallTimeout (sort of call auth) to CGRateS via evapi. Once request sent, we should "park" the request somehow (right now we are considering sleep in the process via usleep and recheck for the answer every 5-10 milliseconds). Answer should be shared via htable. Could you recommend us a better approach? On reply, the answer processed in event route evapi:message-received should set in htable a key of combination callid:fromtag with the maximum call duration or auth reject if not enough balance. This key should be processed by the sleeping process on wakeup. What do u think, is there a better way to approach the 2 ways communication? Thanks in advance for any kind of tip! DanB

2 1

Kamailio TCP Proxy
by Andrew Mason 04 Dec '14

04 Dec '14

Hi All, I am testing SIP messages through the proxy from one client to another and the sender specifies TCP as the transport but Kamailio seems to overwrite this and chooses UDP (TCP is specified in the contact header). The receivers are manually registered using kamctl ul add so i am not sure if this is forcing UDP or something else - I have redirected the messages to wireshark and attached an example of Kamailio adding its VIA headers but specifying UDP (taken from destination). Basically I am getting TCP to Proxy the UDP from Proxy to Dest? Thanks Andrew [image: Inline images 1]

2 1

Some Questions about Kamailio UDP Transport and Concurrency
by pengxuejin＠gangbeng.com 04 Dec '14

04 Dec '14

Dear Kamailio Developers： Hi. I have some questions about kamailio, and want to gain some suggestions from you. 1、The 4 UDP child processes are using system resources(CPU、memory), but they are using only one CPU core(I use 8 core CPU). Why? 2、The ablity of handling the Concurrency is alike, no matter how many UDP child processes I use. Why? And what's the effection of the other UDP child processes? 彭学金 186-2708-0286 技术部开发工程师武汉九午科技有限公司武汉市洪山区光谷大道国际企业中心锦丰楼A座405室 Tel:(027)87108832

2 1

Issue with 4.2.0 and nathelper and/or rtpproxy
by Igor Potjevlesch 04 Dec '14

04 Dec '14

Hello, Since I done the upgrade of Kamailio into 4.2.0, there are, at least, one new case where the RTPProxy is launched. When I got an INVITE from my PSTN Gateway, for unknown reason (and not for all calls), Record-Route is append with nat=yes. I looked into an old trace and this behaviour didn't exist. What could have changed that can explain this? Regards, Igor.

3 16

PSTN Routing with Prefix
by Massimo Varriale (IPZeta) 03 Dec '14

03 Dec '14

Hi, I'm trying to send outbound calls to an external IP address adding a prefix, is this possible using function rewritehostport? Thank you Max

3 3

One way audio calls
by Will Ferrer 03 Dec '14

03 Dec '14

Hi All We are experiencing an odd problem with 1 way audio in about %6-%8 of our calls. It seems to me that what is happening is that our kamailio installation (using rtpengine) is some times ignoring the c= attribute that the carrier is sending us, and sending our audio instead to either address in the carriers o= attribute, or just to the same address we sent our sip invite too. We have looked at hundreds of calls and haven't found a reliable pattern why this happens some times and not others. It really happens in a lot of different scenarios with lots of different carriers and differently formatted SDP packets. I was wondering if any one had an idea where to start for a solution, or a way to make rtpengine always send audio to the ip in the c= attribute we are receiving from the carrier. I haven't been able to find anything so far. Thank you much for any assistance. All the best. Will

2 5

error load "uac.so"
by Javier Ricke 02 Dec '14

02 Dec '14

root@serverjavier-desktop:/home/serverjavier# locate uac.so /usr/local/lib64/kamailio/modules/uac.so /usr/local/src/kamailio-4.0.1/modules/uac/uac.so exists module! but # 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/:/usr/local/lib64/kamailio/mod$ #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif give me a error, if the put # in the loadmodule "uac.so", dont give me error really dont understand

3 4

Noob hoping for guidance
by john＠millican.us 02 Dec '14

02 Dec '14

Hello, I have been googling for a simple (as simple as it can be knowing the complexity of Kamailio) for a "how to/step bey step/do this dummy" type of guide to get Kamailio registered with a sip provider I already have, and register one end point to kamailio that can use that provider. Nothing fancy or involved just the absolute basics. Does this exist and if so can you please point me to it? Thanks, JohnM

3 5

tsilo question
by Klaus Darilion 02 Dec '14

02 Dec '14

Hi! Currently tsilo allows to add branches in failure route or request route (e.g. on REGISTER). would be be possible to allow this function also in reply routes? E.g. if one branch replies with 3xx, to add a new branch to the CFWD target but still have the other branches ringing. regards Klaus

3 2

SDP media IP variable
by Kelvin Chua 02 Dec '14

02 Dec '14

Hi, Is there a way to get the SDP media IP from an INVITE or 183 or 200? and store it in a variable? Kelvin Chua

3 2

how to access request uri of answering branch?
by Juha Heinanen 02 Dec '14

02 Dec '14

lets say that invite is parallel forked to two contacts. if the call is answered, i would like to set in reply route an accounting avp that contains the request uri of answering branch. i have tried $T_req($ru), but that gives the original request uri. any suggestions? -- juha

3 11

Install snmpstats
by Igor Potjevlesch 01 Dec '14

01 Dec '14

Hello, I tried to install the snmpstats module following this process : I installed the dependencies for snmpstats module, then i modified modules.lst. Then i made a make modules, make, and make install. Everything is fine so far. After that, i edited kamailio.cfg the loadmodule for snmstats.so and when i launch kamailio I have the following error : "kamailio: ERROR: <core> [modparam.c:166]: set_mod_param_regex(): No module matching <snmpstats> found" I also made a ldd : # ldd -v -d -r /usr/local/lib64/kamailio/modules/snmpstats.so linux-vdso.so.1 => (0x00007ffff53ff000) libnetsnmpmibs.so.30 => not found libnetsnmpagent.so.30 => not found libnetsnmp.so.30 => not found libkcore.so.1 => /usr/local/lib64/kamailio/libkcore.so.1 (0x00007fca7823b000) libc.so.6 => /lib64/libc.so.6 (0x00007fca77ea8000) /lib64/ld-linux-x86-64.so.2 (0x0000003d6ee00000) undefined symbol: log_color (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: dprint_crit (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: log_prefix_val (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: default_global_address (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: log_stderr (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: process_no (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: core_cfg (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: mem_block (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: default_global_port (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: log_level_info (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: dprint_color_reset (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: parse_params (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: parse_msg (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: fm_malloc (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: get_sock_info_list (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: counter_register (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: parse_uri (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: counter_lookup_str (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: clear_branches (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: get_debug_level (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: my_pid (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: parse_headers (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: dprint_color (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: fm_free (/usr/local/lib64/kamailio/libkcore.so.1) undefined symbol: full_version (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: ver_name (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: log_color (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_child_cb_first (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_global_lock (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_global (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_ginst_count (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: shm_block (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: dprint_crit (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: mem_lock (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: log_prefix_val (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: ver_arch (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_group (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: log_stderr (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_local (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: process_no (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: tcp_disable (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_block_size (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: ver_compiler (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: netsnmp_table_array_helper_handler (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: ver_os (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_child_cb (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: core_cfg (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: ver_flags (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: ver_compiled_time (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: mem_block (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: ver_id (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: ver_version (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: log_level_info (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_child_cb_last (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: dprint_color_reset (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: snmp_get_do_debugging (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: netsnmp_set_mode_request_error (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: register_timer (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: netsnmp_ds_set_boolean (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_get_by_name (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: send_v2trap (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: netsnmp_table_array_check_row_status (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: netsnmp_register_scalar (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: tcp_options_get (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: netsnmp_container_find (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: fm_malloc (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_register_child (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: init_agent (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: register_procs (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: debugmsg (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: get_ticks (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: netsnmp_check_vb_rowstatus (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_register_ctx (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: debugmsgtoken (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: init_snmp (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: counter_lookup_str (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: get_debug_level (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_child_init (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: snmp_set_var_typed_value (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: snmp_free_varbind (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: parse_oid_indexes (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: netsnmp_table_container_register (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: netsnmp_create_handler_registration (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: agent_check_and_process (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: snmp_shutdown (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: find_module_by_name (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: fork_process (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: netsnmp_check_vb_type_and_size (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: cfg_move_handle (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: counter_get_val (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: snmp_varlist_add_variable (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: my_pid (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: snmp_reset_var_buffers (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: fm_info (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: snmp_log (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: dprint_color (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: snmp_clone_mem (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: fm_free (/usr/local/lib64/kamailio/modules/snmpstats.so) undefined symbol: find_export (/usr/local/lib64/kamailio/modules/snmpstats.so) Version information: /usr/local/lib64/kamailio/modules/snmpstats.so: libc.so.6 (GLIBC_2.2.5) => /lib64/libc.so.6 /usr/local/lib64/kamailio/libkcore.so.1: libc.so.6 (GLIBC_2.3) => /lib64/libc.so.6 libc.so.6 (GLIBC_2.3.4) => /lib64/libc.so.6 libc.so.6 (GLIBC_2.2.5) => /lib64/libc.so.6 /lib64/libc.so.6: ld-linux-x86-64.so.2 (GLIBC_PRIVATE) => /lib64/ld-linux-x86-64.so.2 ld-linux-x86-64.so.2 (GLIBC_2.3) => /lib64/ld-linux-x86-64.so.2 Can you help me with this please ? By the way I configured /etc/snmpd.conf by adding the line "master agentx" and created a file /etc/snmpstats.cfg with only one line : "agentXSocket tcp:localhost:705". Is that correct ? Do i need to reload something after modifing those files ?

1 1

kamailio error, failed to start.
by Javier Ricke 01 Dec '14

01 Dec '14

root@serverjavier-desktop:/usr/local/etc/kamailio# /etc/init.d/kamailio restart Restarting Kamailio: loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: 152.74.21.12:5060 tcp: 127.0.0.1:5060 tcp: 152.74.21.12:5060 Aliases: tcp: localhost:5060 udp: localhost:5060 kamailio error, failed to start. i dont know what happen plz!

3 5

Paid Consultancy
by Massimo Varriale (IPZeta) 01 Dec '14

01 Dec '14

Hi Guys! As I didn't got any answer from my past question regarding Carrier Route module implementation/routing, maybe I will get answers if I will ask my company to pay for a support. Is there a paid service for this? Thanks Max

2 1

Kamailio as pass through proxy?
by Rizwan Khan 01 Dec '14

01 Dec '14

Hi All, What we want to do is to by-pass the restrictions imposed by different ISPs etc. which normally block based on the port 5060 The SIP server(including SBC) is a commercial one running on port 5060, that we cannot change. It is already taking care of the NAT issues. We want to use Kamailio as the pass-thru proxy in front of the SBC just to receive the request on a random port, and then forward the request to the SBC internally being on the same private network. What would be the best way to do it? Any help will be highly appreciated. Rizwan Khan

3 3

Error when using function lookup_branches
by J Hazenberg 30 Nov '14

30 Nov '14

Hello, I'm runnig kamailio 4.2.0 and using the lookup_branches function from the register module to lookup all branches after a alias_db_lookup. The function seems to work fine but i see the following error in the log file just after the lookup: INFO: <core> [mem/f_malloc.c:599]: fm_free(): freeing a free fragment (0x7fad277fe110/0x7fad277fe148) - ignore INFO: <core> [mem/f_malloc.c:599]: fm_free(): freeing a free fragment (0x7fad277fcd38/0x7fad277fcd70) - ignore INFO: <core> [mem/f_malloc.c:599]: fm_free(): freeing a free fragment (0x7fad277fcd88/0x7fad277fcdc0) - ignore Code: alias_db_lookup("dbaliases"); if (!lookup_branches("location")) { This only happens when i have multiple branches, could this be a error in my script? Thanks, Jan

4 13

New Socket Per Call
by Andrew Mason 29 Nov '14

29 Nov '14

Hi All, I am looking to emulate a local proxy with Kamailio or OpenSips using a UAC and a UAS (each on separate machines ) but when initiate traffic from UAC to UAS using SIPp and defining a new socket for each call it does not force this between proxy and UAS but does between UAC and proxy - I think this may be a routing configuration? Thanks Andrew

1 0

[Kamailio 4.2.0] Carrieroute Routing Questions
by Massimo Varriale (IPZeta) 29 Nov '14

29 Nov '14

Hi Guys, I'm new on Kamailio and this list, so be patient with me :) I've built an almost "complete" working SIP server on Ubuntu 14.04 LTS. I told almost complete because my problem is with carrierroute module because I'm not understanding the routing file and where I should put the code to use function cr_route I would like to allow SIP calls/video between users for free and send calls to different external Gateways / Switches, etc based on the dialled destination. In particular, I can send outbound calls a prepaid platform that will allow to bill calls based on CLI validation, but in all my tests, it seems that calls are going out always to the IP address of the remote gateway found into the carrierroute table ignoring the SIP user I'm calling.....what is the correct routing? I'm finding some docs around there, but some refer to very old versions of Kamailio and some others are not clear to me. What's wrong with my routing logic? Thank you so much! Max route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # 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); # CARRIERROUTE MODULE routing logic # check table for carrier default and domain default if(!cr_route("default", "default", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { # In case of failure, re-route the request t_on_failure("1"); # Relay the request to the gateway t_relay(); } # user location service route(LOCATION); route(RELAY); }

1 2

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

28 Nov '14

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

6 12

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

28 Nov '14

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

2 3

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

28 Nov '14

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

2 1

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

28 Nov '14

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

1 0

CNXCC PrePaid module
by Rene Montilva 27 Nov '14

27 Nov '14

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

3 16

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

27 Nov '14

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

3 2

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

27 Nov '14

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

1 0

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

27 Nov '14

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

1 0

Dispatcher BYE issue
by Ahmed Salem 26 Nov '14

26 Nov '14

Hi, I'm trying to achieve this configuration phone --> NAT --> kamailio loadbalancer --> kamailio sip/proxy --> kamailio sip/proxy With the attached configuration I can make calls without any problems but when the callee hangs up the call does not end only the caller can hang up. Btw I'm using tcp for signalling. Any ideas? Thanks #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 mhomed=1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 tcp_async=yes tcp_connection_lifetime=3605 port=5060 sip_warning=yes ####### Modules Section ######## #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!endif loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "dispatcher.so" loadmodule "nathelper.so" loadmodule "rtpproxy.so" loadmodule "path.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 0) # ----- tm params ----- modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 3000) modparam("tm", "fr_inv_timer", 120000) # ----- dispatcher params ----- modparam("dispatcher", "db_url", "mysql://kamailio:kamailiorw@localhost/kamailio") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) #modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "flags_col", "flags") modparam("dispatcher", "priority_col", "priority") modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("path", "use_received", 1) modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) ####### Routing Logic ######## # main request routing logic request_route { xlog("L_NOTICE", "*** loose routing enforced before: $rm, $ru, $si, $du\n"); # per request initial checks route(REQINIT); loose_route(); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers #remove_hf("Route"); if (is_method("INVITE")){ record_route(); } if(is_method("REGISTER")){ add_path_received(); fix_nated_register(); } else{ add_contact_alias(); } # dispatch destinations route(DISPATCH); route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("L_NOTICE", "*** loose routing enforced before: $rm, $ru, $si, $du\n"); xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests #route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } sl_send_reply("404","Not here"); } exit; } } # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_domain("1", "0")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); return; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_domain()) { xlog("L_NOTICE", "going to next dst\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } }

2 2

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

25 Nov '14

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

4 9

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

25 Nov '14

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

3 2

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

25 Nov '14

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

3 2

sqlite load module error
by Javier Ricke 24 Nov '14

24 Nov '14

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

2 2

host undefined
by Javier Ricke 24 Nov '14

24 Nov '14

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

3 4

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

24 Nov '14

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

2 1

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

24 Nov '14

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

3 2

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

24 Nov '14

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

2 1

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

21 Nov '14

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

2 2

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

21 Nov '14

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

1 0

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

20 Nov '14

Hello! I'm having this issue and I'm not able to solve it. I have searched it many times on the Internet but I found nothing, that's why I am asking here... I'm sorry I do not wanna cause any troubles but I need to get this done... This is my scenario, I want to make a call through a Kamailio server that redirects to another server. I have an User A registered to Server A and a User B registered to Server B, what I want to do is the following: User A calls User B using Server A, because Server A doesn't have the User B the Server A will redirect to Server B and make the call. I'm using Ekiga and of course Kamailio (and I want to make it work both ways). This is the code I'm using to do this in kamailio.cfg, I replaced this lines of code: *$avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } }* For this ones: *$avp(oexten) = $rU; if (!lookup("location")) { sl_send_reply("300",("REDIRECT")); rewritehostport("10.80.129.177"); forward(uri:host,uri:port); break; }* And it didn't work, then I putted the lines above as the following: *$avp(oexten) = $rU; if (!lookup("location")) { * * forward(uri:host,uri:port); rewritehostport("10.80.129.177");* * sl_send_reply("300",("REDIRECT")); break; }* And it worked but if I do a Wireshark Capture on the Server I get a lot of REDIRECT packets and my teacher says this is wrong, that I should only get 1. He did not help me and said I must learn and I want but I'm not able to understand what's going on, I need to do this because I want to pass the exam and not to fail... Again I'm sorry for taking your time, I'm asshamed. Thank you very much

2 1

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

20 Nov '14

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

2 1

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

19 Nov '14

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

3 5

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

19 Nov '14

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

2 6

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

19 Nov '14

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

2 1

Training/Tutorials?
by Jordan Rinke 19 Nov '14

19 Nov '14

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

2 1

rtpengine UDP/TLS/RTP/SAVP <=> RTP/AVP re-INVITE
by Juha Heinanen 19 Nov '14

19 Nov '14

when i make call from UDP/TLS/RTP/SAVP baresip to RTP/AVP sems, rtpengine gets called on initial invite/200 ok like this and audio works fine: Nov 18 02:46:06 rautu /usr/bin/sip-proxy[926]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1 RTP/AVP trust-address) Nov 18 02:46:06 rautu /usr/bin/sip-proxy[878]: INFO: ===== rtpengine_answer(ICE=force via-branch=2 trust-address) however, when baresip makes re-invite, rtpengine gets called using the same offer/answer flags, but sems clears the call. on syslog, i see this: Nov 18 02:46:59 rautu /usr/bin/sip-proxy[919]: INFO: Routing in-dialog INVITE <sip:127.0.0.1:5090;transport=udp> from <sip:jh@test.tutpro.com> Nov 18 02:46:59 rautu /usr/bin/sip-proxy[879]: INFO: ===== rtpengine_answer(ICE=force via-branch=2 replace-session-connection replace-origin) Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50444] SRTP output wanted, but no crypto suite was negotiated Nov 18 02:46:59 rautu /usr/bin/sip-proxy[919]: INFO: Routing in-dialog ACK <sip:127.0.0.1:5090;transport=udp> from <sip:jh@test.tutpro.com> Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50462] Discarded invalid SRTP packet: authentication failed Nov 18 02:46:59 rautu sems[20439]: [#b7193b70] [receive, AmRtpAudio.cpp:212] ERROR: decode() returned -4 Nov 18 02:46:59 rautu /usr/bin/sip-proxy[878]: INFO: ===== rtpengine_delete() Nov 18 02:46:59 rautu /usr/bin/sip-proxy[878]: INFO: Routing in-dialog BYE <sip:jh-0x9a95b00@192.98.102.30:5066;transport=tcp> from <sip:jh@as.test.tutpro.com> to <sip:192.98.102.30:46718;transport=TCP> based on gruu Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50463] Error parsing RTCP header: invalid packet type and on baresip console this: dtls_srtp: ---> DTLS-SRTP complete (audio/RTCP) Profile=AES_CM_128_HMAC_SHA1_80 dtls_srtp: incoming DTLS connect from 192.98.102.30:50524 dtls_srtp: verified sha-1 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 srtp: recv: failed to decrypt RTCP-packet (Unknown error 217) srtp: recv: failed to decrypt RTCP-packet (Unknown error 217) srtp: recv: failed to decrypt RTP-packet (Unknown error 217) srtp: recv: failed to decrypt RTP-packet (Unknown error 217) srtp: recv: failed to decrypt RTP-packet (Unknown error 217) srtp: recv: failed to decrypt RTP-packet (Unknown error 217) sip:jh@as.test.tutpro.com: session closed: Connection reset by peer could it be that at some point during the re-invite, sems gets srtp audio and therefore clears the call? if so, is it a bug in rtpengine? -- juha

2 2

reREGISTER problems: contact is inserted instead updated
by Klaus Darilion 18 Nov '14

18 Nov '14

Hi! Kamailio 4.1.6 with default settings for usrloc and registrar modul and db_mode=1 (write through). Fritzbox sends a reREGISTER and Kamailio tries to insert the contact into the database, but this fails as this contact for this AoR already exists. So I think there are 2 scenarios: Kamailio false INSERTs instead of UPDATEs the contact, or the contact was deleted from usrloc, but not from the database. Frankly I see the in the mysql logfile that immediately after the failed INSERT the record is deleted from DB. Does anybody have a pointer what might be the problem here? AFAIS the client always uses the same Contact and the same call-id. only Cseq increases. Thanks Klaus

1 1

rtpengine flags UDP/TLS/RTP/SAVP and UDP/TLS/RTP/SAVPF
by Juha Heinanen 18 Nov '14

18 Nov '14

rtpengine README lists these rtp protocol and profile flags: + RTP/AVP, RTP/SAVP, RTP/AVPF, RTP/SAVPF rtpengine source code, however, seems to know also these DTLS ones: daemon/call.c: .name = "UDP/TLS/RTP/SAVP", daemon/call.c: .name = "UDP/TLS/RTP/SAVPF", can they be used in rtpengine_offer/answer calls even when they are missing from README? -- juha

2 3

Re: [SR-Users] UDP-TCP bridging in-dialog SUBSCRIBE
by Dan Christian Bogos 17 Nov '14

17 Nov '14

Hey Paul, Thanks for your feedback! In this case the call will come in via UDP (so it has 2 record route headers). I can still play tricks like reading the record-route headers but, I was in some way hoping that is the responsibility of loose_route() function to properly use the right protocol out of route headers since we are a proxy. Cheers, DanB On 14.11.2014 12:00, sr-users-request(a)lists.sip-router.org wrote: > Message: 3 > Date: Thu, 13 Nov 2014 16:55:18 +0530 > From: Varghese Paul<varghesepaul87(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] UDP-TCP bridging in-dialog SUBSCRIBE > Message-ID: > <CAM33Ch4furD1FM-ZNKfbQ0nXNQd88BmzW73gjjb=cVyuC0K=8w(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi Dan, > > You can check the transport value in the RURI and relay the SIP message as > tcp. > > if ($*rP* == "tcp"){ > t_relay_to_tcp() > } > > Regards > > Varghese Paul

3 4

Siremis Installation Issues
by Daniel Shoemaker 17 Nov '14

17 Nov '14

This is my first time posting to the mailing list. I'm currently running Kamailio 4.1, per the instructions on the website I got the tar bal for Siremis 4.1 s well. I have followed the install instructions at http://kb.asipto.com/siremis:install41x:main and I get all the way down to the bottom where I need to login to the web interface and when I try to use the default I get a very length openbiz error: [{"target":"ERROR","content":" \n[2014-11-17 11:51:15 (GMT)] An exception occurred while executing this script: \nError message: #8192, Non-static method BizSystem::dbConnection() should not be called statically, assuming $this from incompatible context<\/font> \nScript name and line number of error: \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php:423<\/font> \n *function:<\/b> errorHandler ( 8192, \"Non-static method BizSystem::dbConnection() should not be called...\", \"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\", 423, Array(0) ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/sysheader.inc 117\nfunction:<\/b> userErrorHandler ( 8192, \"Non-static method BizSystem::dbConnection() should not be called...\", \"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\", 423, Array(0) ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php 423\nfunction:<\/b> getDBConnection ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Lite.php 406\nfunction:<\/b> _run_search ( Array(2) ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Lite.php 345\nfunction:<\/b> fetchRecords ( \"[username]='admin' and status='1'\", Array(0), 1 ) @ \/var\/www\/siremis-4.1.0\/siremis\/modules\/service\/authService.php 98\nfunction:<\/b> authDBUser ( \"admin\", \"admin\" ) @ \/var\/www\/siremis-4.1.0\/siremis\/modules\/service\/authService.php 66\nfunction:<\/b> authenticateUser ( \"admin\", \"admin\" ) @ \/var\/www\/siremis-4.1.0\/siremis\/modules\/user\/form\/LoginForm.php 92\nfunction:<\/b> Login ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php 310\nfunction:<\/b> invoke ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php 110\nfunction:<\/b> dispatchRequest ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php 32\nfunction:<\/b> include_once ( \"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php\" ) @ \/var\/www\/siremis-4.1.0\/siremis\/bin\/controller.php 6<\/div>\n------------------------------Please ask system administrator for help...<\/div>\n"}]* My web server is configured to use apache and I've checked the ownership on the files and folder. I did a chmod 775 starting at the siremis-4.1.0/ directory. I did a search and found that the openbiz folder needs to be right under the siremis-4.1.0/ folder which it is. I have tried reinstalling several times only to get the same result. I'm sure I'm missing something simple I just need to know what it is. I'm currently running CentOS 7, Kernel 3.10.0-123.9.3, php 5.6.2-3, mariadb 10.0.14-1, & httpd 2.4.6-18. Dan --

1 0

Kamailio TCP Problems
by Andrew Mason 17 Nov '14

17 Nov '14

Hi All, Looking for some help with a TCP issue in Kamailio. I am running load testing through the proxy with a topology of UAC > Proxy > UAS. With UDP it works fine (using SIPp scripts) - when I turn to TCP as the transport i get the following error in the kamailio logs: ERROR: <core> [tcp_main.c:4159]: handle_tcpconn_ev(): connect 192.168.0.19:5060 failed The IP address stated is the UAC which is sending an INVITE to the Proxy, which seems to send back a 100 TRYING to the UAC which is followed by repeated INVITE messages. It looks like the leg from the Proxy to the UAS is never established using TCP - however if I send the SIP messages to the Proxy via TCP and set the UAS to receieve in UDP Kamailio completes with TCP > Kamailio > UDP. When the UAC is set to send on TCP and UAS set to receive on TCP netstat shows the ports as listening. Can anyone please help?

1 0

Kamailio 4.2 TLS and csipsimple
by nadie 16 Nov '14

16 Nov '14

Hello. I installed K 4.2 with TLS. I can login users with csipsimple in TLS. but users can't call to other. in 5060 all correct. Thanks.

1 0

kamailio with pipelimit crashes when CPU load average >2
by Julia Boudniatsky 16 Nov '14

16 Nov '14

Hello, Sometimes kamailio with pipelimit crashes with segmentation fault, when CPU load average >2. The problem is not reproduced in the LAB (we used stress –c ). Please find attached core file. Thank you, Julia

1 0

401 vs 407 UAC auth issue
by Satish Patel 15 Nov '14

15 Nov '14

We have opensip configured with UAC auth to register SIP provide. I have configured two provider and both got registered but very interesting thing happened. Provider "A" sending me 407 challenge for authentication - Working Provider "B" sending me 401 for authentication - its failed to auth what is the solution here? does UAC module work with 401?

1 1

Secure WebSocket connection is getting dropped
by Kamrul Khan 14 Nov '14

14 Nov '14

Hi, I have a wired problem in my Kamailio. Over WSS, Kamailio is dropping the websocket connection right after it is getting the BYE from other side. So the client side is not getting any BYE. Over normal http it is working fine. Any idea why it might happen or where to look at? Thanks in advanced!

1 0

Non standard TLS port
by elektau＠yandex.ru 14 Nov '14

14 Nov '14

Hello All, I'm using Kamailio 4.2 running TLS only connection on non standard port 5065. I added the following line to the kamailio.cfg: listen=tls:111.222.333.444:5065 Domain SRV record as follows: _sips._tls.domain.com port 5065 When I try to connect using a softphone to domain.com login/password - "error while registering" When I connect using domain.com:5065 login/password - connection established successfully. So the question is: why I don't get redirected automatically to the specified port? If I use the default port for TLS (5061) in kamailio.cfg and in SRV, I can successfully connect using domain.com login/password Please help. Many Thanks!

4 5

cannot register UA when domain is not valid, using usrloc, auth_db and registrar modules
by Vik Killa 14 Nov '14

14 Nov '14

Hello, I'm trying to register 1004(a)3.local (yes i know 3.local is not a valid domain) and kamailio is responding with SIP/2.0 478 Unresolvable destination (478/SL) Why does kamailio require a valid domain? Is there any way to disable kamailio from looking up the domain (i don't see a reason to look it up since it's just a REGISTER request) Here is my cfg: if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "1"); exit; } And kamilio is responding with: Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6931]: ERROR: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6931]: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6931]: ERROR: tm [t_fwd.c:1778]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6931]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6932]: ERROR: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6932]: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6932]: ERROR: tm [t_fwd.c:1778]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6932]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6933]: ERROR: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6933]: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6933]: ERROR: tm [t_fwd.c:1778]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6933]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Here is SIP ----------------begin of packet ----------------------------- U 2014/11/13 16:37:36.742489 192.168.207.162:63582 -> 192.168.207.156:5060 REGISTER sip:3.local SIP/2.0 Via: SIP/2.0/UDP 192.168.16.120:63582 ;rport;branch=z9hG4bKPjbea638d5e7054e03bc4dfee6b6872275 Max-Forwards: 70 From: "1004" <sip:1003@3.local>;tag=ff09e20e549b4127a88ae6a6ec34e88b To: "1004" <sip:1003@3.local> Contact: <sip:98524760@192.168.16.120:63582 >;+sip.instance="<urn:uuid:7d4ed3bd-451a-4406-a8e2-80315408b7c5>" Call-ID: a7454fb3539740ffb385c146303a926f CSeq: 1 REGISTER Expires: 600 Supported: gruu User-Agent: Blink 0.9.1.2 (Windows) Content-Length: 0 ------------------end of packet ----------------------------- ----------------begin of packet ----------------------------- U 2014/11/13 16:37:36.743505 192.168.207.156:5060 -> 192.168.207.162:63582 SIP/2.0 478 Unresolvable destination (478/SL) Via: SIP/2.0/UDP 192.168.16.120:63582 ;rport=63582;branch=z9hG4bKPjbea638d5e7054e03bc4dfee6b6872275;received=192.168.207.162 From: "1004" <sip:1003@3.local>;tag=ff09e20e549b4127a88ae6a6ec34e88b To: "1004" <sip:1003@3.local>;tag=98beed43ea484d57160c0a87164dfce1.ff17 Call-ID: a7454fb3539740ffb385c146303a926f CSeq: 1 REGISTER Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 ------------------end of packet ----------------------------- ----------------begin of packet ----------------------------- U 2014/11/13 16:37:36.743625 192.168.207.156:5060 -> 192.168.207.162:63582 SIP/2.0 478 Unresolvable destination (478/TM) Via: SIP/2.0/UDP 192.168.16.120:63582 ;rport=63582;branch=z9hG4bKPjbea638d5e7054e03bc4dfee6b6872275;received=192.168.207.162 From: "1004" <sip:1003@3.local>;tag=ff09e20e549b4127a88ae6a6ec34e88b To: "1004" <sip:1003@3.local>;tag=b9d45a0d611ee9eea95e269f12755a03-ff17 Call-ID: a7454fb3539740ffb385c146303a926f CSeq: 1 REGISTER Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 ------------------end of packet ----------------------------- ----------------begin of packet ----------------------------- U 2014/11/13 16:37:36.750939 192.168.207.156:5060 -> 192.168.207.162:63582 SIP/2.0 478 Unresolvable destination (478/SL) Via: SIP/2.0/UDP 192.168.16.120:63582 ;rport=63582;branch=z9hG4bKPj186bbf75b3964f06893742b853533302;received=192.168.207.162 From: "1004" <sip:1003@3.local>;tag=8b337cc975ab41c3b431a335a898128a To: "1004" <sip:1003@3.local>;tag=98beed43ea484d57160c0a87164dfce1.10fc Call-ID: fb19ff0b98f542849a249cd96dcd5f8c CSeq: 1 PUBLISH Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0

2 2

Allow Re-invite mid call
by Eric Koome 13 Nov '14

13 Nov '14

How do I process re-invite from sip provider mid call which check connection. At the moment Kamailio replies 404 not here, and call is dropped. Eric

2 3

avoid OPTION requests without authentication
by Julien 13 Nov '14

13 Nov '14

Hello, I received in my SIP clients an OPTION request forwarded by kamailio. The user agent of this request is 'sipvicious'. I want to avoid these kind of requests from unauthenticated users How can I do that ? -- Julien Barreau

2 3

Re: [SR-Users] UDP-TCP bridging in-dialog SUBSCRIBE
by Dan Christian Bogos 13 Nov '14

13 Nov '14

Thanks Alex for your confirmation. In some way I was afraid about this ;). Have a good one! DanB

5 11

Kamailio - group presence
by Marino Mileti 13 Nov '14

13 Nov '14

Hi guys, I've to servers K1 e K2. On K2 I've registered two users on K1 and 2 users on K2. I would like that a user on K1 can subscribe to status of K2.... when there's a call on K2, it has to publish "On-call" status; if there isn't any ongoin call it has to publish "Online" status. I've tried to use uac module and send "PUBLISH" from scripts and it works but there are a lot of issue with SIP-If-Match header..so on K1 presentity table on db everytime a row is added and the old will be deleted when expires. I've tried also with PUA_MI...but using fifo on K2...so user on K1 should subscribe for a "virtual" user on K2. This isn't a good idea in case of network problem. Is there anyother way or do you have any suggests? ----- Marino Mileti -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-group-presence-tp132512.ht… Sent from the Users mailing list archive at Nabble.com.

1 0

messages delivered by MSILO are not cleared immediately
by Prasun Bheri 13 Nov '14

13 Nov '14

Hi Group, I have configured MSILO with following modparams. modparam("msilo", "db_url",DBURL) modparam("msilo", "offline_message","$rU is offline!") modparam("msilo", "extra_hdrs", "Msilo-msg: true\r\n") modparam("msilo", "outbound_proxy", MSILO_OUTBOUND_PROXY) modparam("msilo", "content_type_hdr","Content-Type: message/cpim\r\n") modparam("msilo", "extra_hdrs", "Content-Type: message/cpim\r\n") modparam("msilo", "add_date", 0) MSILO dumps all the messages to destination immediately after recipient registers and gets 200 ok for sent messages. The delivered messages are not cleared immediately and takes a minute to delete each message. Could some one explain why the delay is observed? In case you need to know I am using Kamailio 4.0.0 on CentOS 6.5 Thanks & Regards, Prasun

1 0

Kamailio Routing Logic
by Zaka Ul Isam 13 Nov '14

13 Nov '14

Hello Friends and Gurus: I am struggling with routing and need a bit of clarification WRT following: route { if ($fU=="601"&&is_method("INVITE")) record_route(); rewritehost("10.0.10.111"); //SS IP route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } * In the above stateful forwarding what is the rationale of calling route (1) within main route block? Could the t_relay statement afyer rewriting the host part of URI not have performed the stateful forwarding? ** In route[1] there is is only an expression abt logical not condition about t_relay() (if methods in this function are not invoked??) I wonder where exactly is stateful forwarding taking place? 1- Is it the default action? 2- Or in function sl_reply_error()? Thanks a lot in advance! Zaka ________________________________ Albtelecom & Eagle Mobile invite you to protect together the Environment. Please do not print the e-mail unless really necessary. Albtelecom & Eagle Mobile iu ftojnë të mbrojmë së bashku Mjedisin. Lutemi të mos e printoni këtë komunikim elektronik nëse nuk është vërtet i nevojshëm. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the addressee/s. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited. If you receive this e-mail in error, please notify the sender immediately and delete it! Albtelecom makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message may belong to sender alone and may not necessarily reflect the opinions of Albtelecom. Albtelecom shall bear no liability for any loss or damage caused by software or e-mail viruses. Ky mesazh dhe çdo informacion i transmetuar në përmbajtje te këtij mesazhi është konfidencial dhe është i destinuar vetëm për marrësin e destinuar. Nëse nuk jeni marrësi i destinuar, Ju bëjmë me dije se çdo përhapje, transmetim, kopjim apo përdorim i çdo informacioni është i ndaluar. Nëse e merrni këtë mesazh gabimisht, ju lutem kontaktoni urgjentisht nisësin e tij dhe fshijeni atë. Albtelecom nuk jep asnjë garanci për saktësinë apo plotësinë e informacionit në përmbajtje të këtij mesazhi dhe nuk mban asnjë përgjegjësi për informacionin e përmbajtur, transmetimin, marrjen, ruajtjen apo përdorimin e tij në çfarëdolloj mënyre. Mendimet e shprehura në këtë mesazh mund t'i përkasin vetë nisësit dhe nuk mund të reflektojnë domosdoshmërisht qëndrimet e Albtelecom. Albtelecom nuk do të mbaje asnjë përgjegjësi për humbje ose dëme te shkaktuara nga programet apo viruset.

2 4

result of param.value transformation when param does not exist?
by Juha Heinanen 13 Nov '14

13 Nov '14

wiki transformations page on param.value does not mention, what is the result of the transformation when named param does not exist. same with many other transformations. what is the value if referenced value does not exit? should it be mentioned on the wiki page? -- juha

2 4

topoh and uac_replace_from()
by Alex Balashov 13 Nov '14

13 Nov '14

Hi, If I use uac_replace_from() to alter the User part of the From URI: uac_replace_from("sip:$xavp(call=>src)@$fd"); it overwrites the domain concealment that is otherwise imparted by topoh, i.e. restores the original From domain. However, uac_replace_from() does not appear to offer the option to provide a replacement user part only, so I have to specify the domain. As far as I know, there is no means of accessing the modifications topoh has made, as this is transparent from the point of view of the config. Any way around this problem? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 3

kamailio does DNS lookup of from host when using TCP
by Vik Killa 12 Nov '14

12 Nov '14

Hello, I'm wondering why kamailio will try doing a DNS lookup of the from host with a REGISTER packet when using TCP? It doesn't do this with UDP. I always get this error since the host is not a valid domain: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "3.local" Here is the full log with debug (error is last line) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [tcp_main.c:2294]: tcpconn_send_put(): tcp_send: found fd in cache ( 10, 0x7f860bd18938, 1) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f860bd18938 n=639 fd=10 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf=#012SIP/2.0 401 Unauthorized#015#012Via: SIP/2.0/TCP 192.168.16.120:53972;received=192.168.213.162;rport=53972;branch=z9hG4bKPjac894585917043d28da015d397178725;alias#015#012From: "1003" <sip:1003@3.local>;tag=dbc7a9087cb84006b65d891e5ad1f5ea#015#012To: "1003" <sip:1003@3.local>;tag=ey7D0K9DX49DF#015#012Call-ID: ee72194a1b8e4effa2948e237d804b90#015#012CSeq: 1 REGISTER#015#012User-Agent: AMPSwitch#015#012Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE#015#012Supported: timer, path, replaces#015#012WWW-Authenticate: Digest realm="3.local", nonce="cdf32dde-ab14-4265-b491-bee703e83255", algorithm=MD5, qop="auth"#015#012Content-Length: 0#015#012#015#012 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: tm [t_reply.c:1663]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [receive.c:298]: receive_msg(): receive_msg: cleaning up Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0xa2c420, 6, 2, 0x7f860bd2dbe0), fd_no=1 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 235, <rport> = <n/a>; state=6 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKPj19019d19feaa4a06ac1d6cdc31a2e05d>; state=6 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 237, <alias> = <n/a>; state=16 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg... Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [receive.c:197]: receive_msg(): preparing to run routing scripts... Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=10 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [27]; uri=[sip:1003@3.local] Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body ["1003" <sip:1003@3.local>#015#012] Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <1> <PUBLISH> Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=757 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: textops [textops.c:2478]: has_body_f(): content type is 196613 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: siputils [checks.c:103]: has_totag(): no totag Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=10 global id=9 T start=0xffffffffffffffff Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=12001, isACK=0 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=10 global id=10 T end=(nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:1373]: t_newtran(): DEBUG: t_newtran: msg id=10 , global msg id=10 , T on entrance=(nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=12001, isACK=0 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [md5utils.c:67]: MD5StringArray(): DEBUG: MD5 calculated: d71ddc300035dfaa15e181fe4d6a9aa7 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [dns_cache.c:566]: _dns_hash_find(): dns_hash_find(_sip._tcp.3.local(17), 33), h=867 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [resolve.c:770]: get_record(): get_record: lookup(_sip._tcp.3.local, 33) failed Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [dns_cache.c:894]: dns_cache_mk_bad_entry(): dns_cache_mk_bad_entry(_sip._tcp.3.local, 33, 60, 1) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [dns_cache.c:827]: dns_cache_add(): dns_cache_add: adding _sip._tcp.3.local(17) 33 (flags=1) at 867 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [dns_cache.c:566]: _dns_hash_find(): dns_hash_find(3.local(7), 1), h=184 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [resolve.c:770]: get_record(): get_record: lookup(3.local, 1) failed Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [dns_cache.c:894]: dns_cache_mk_bad_entry(): dns_cache_mk_bad_entry(3.local, 1, 60, 1) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [dns_cache.c:827]: dns_cache_add(): dns_cache_add: adding 3.local(7) 1 (flags=1) at 184 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: ERROR: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "3.local"

2 4

UDP-TCP bridging in-dialog SUBSCRIBE
by Dan Christian Bogos 12 Nov '14

12 Nov '14

Hey Guys, I have the following scenario: UA1 (UDP) -> Kamailio Proxy -> (TCP) UA2 As per the signaling sample, I would expect that in-dialog SUBSCRIBEs are bridged between UDP and TCP network but for some reason this does not happen (the subscribe comes in over UDP and leaves still over UDP although tcp is present in the route headers). Are we somehow forced to mangle contact header and add transport parameter to it or the record-route headers should serve for this purpose? Any tip would be highly appreciated. Ta, DanB Bellow the traces for initial SUBSCRIBEs as well as in-dialog ones (info mangled to protect privacy). ##################################### Normally routed request (non-dialog one, correctly bridged) ##################################### """ # U 2014/11/12 19:04:51.293887 pub_ip_ua:54705 -> pub_ip_ep:5060 SUBSCRIBE sip:user1@fakedomain.com:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.243.82:5062;branch=z9hG4bK1114302228. From: "user2" <sip:user2@fakedomain.com>;tag=1003326118. To: <sip:user1@fakedomain.com>. Call-ID: 1273081718. CSeq: 2 SUBSCRIBE. Contact: <sip:user2@192.168.243.82:5062>. Accept: application/dialog-info+xml. Max-Forwards: 70. User-Agent: Tiptel IP 284 6.70.13.18 0015651b574f. Expires: 60. Event: dialog. Content-Length: 0. . # T 2014/11/12 19:04:51.295706 pub_ip_ep:34052 -> 192.168.51.168:5060 [AP] SUBSCRIBE sip:user1@192.168.51.168:5060;transport=tcp SIP/2.0. Record-Route: <sip:pub_ip_ep;transport=tcp;r2=on;lr;nat=yes>. Record-Route: <sip:pub_ip_ep;r2=on;lr;nat=yes>. Via: SIP/2.0/TCP pub_ip_ep;branch=z9hG4bKe26f.3dce66bd1976447b88f6096e926ce68c.0. Via: SIP/2.0/UDP 192.168.243.82:5062;rport=54705;received=pub_ip_ua;branch=z9hG4bK1114302228. From: "user2" <sip:user2@fakedomain.com>;tag=1003326118. To: <sip:user1@fakedomain.com>. Call-ID: 1273081718. CSeq: 2 SUBSCRIBE. Contact: <sip:user2@192.168.243.82:5062;alias=pub_ip_ua~54705~1>. Accept: application/dialog-info+xml. Max-Forwards: 65. User-Agent: Tiptel IP 284 6.70.13.18 0015651b574f. Expires: 60. Event: dialog. Content-Length: 0. . # T 2014/11/12 19:04:51.300350 192.168.51.168:5060 -> pub_ip_ep:34052 [AP] SIP/2.0 202 Accepted. Via: SIP/2.0/TCP pub_ip_ep;branch=z9hG4bKe26f.3dce66bd1976447b88f6096e926ce68c.0;rport=34052. Via: SIP/2.0/UDP 192.168.243.82:5062;rport=54705;received=pub_ip_ua;branch=z9hG4bK1114302228. Record-Route: <sip:pub_ip_ep;transport=tcp;r2=on;lr;nat=yes>. Record-Route: <sip:pub_ip_ep;r2=on;lr;nat=yes>. From: "user2" <sip:user2@fakedomain.com>;tag=1003326118. To: <sip:user1@fakedomain.com>;tag=PFxfnedC0jP7. Call-ID: 1273081718. CSeq: 2 SUBSCRIBE. Contact: <sip:user1@192.168.51.168:5060>. Expires: 60. User-Agent: Sipean-AS 1.2.10. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: active;expires=60. Content-Length: 0. . ## U 2014/11/12 19:04:51.300745 pub_ip_ep:5060 -> pub_ip_ua:54705 SIP/2.0 202 Accepted. Via: SIP/2.0/UDP 192.168.243.82:5062;rport=54705;received=pub_ip_ua;branch=z9hG4bK1114302228. Record-Route: <sip:pub_ip_ep;transport=tcp;r2=on;lr;nat=yes>. Record-Route: <sip:pub_ip_ep;r2=on;lr;nat=yes>. From: "user2" <sip:user2@fakedomain.com>;tag=1003326118. To: <sip:user1@fakedomain.com>;tag=PFxfnedC0jP7. Call-ID: 1273081718. CSeq: 2 SUBSCRIBE. Contact: <sip:user1@192.168.51.168:5060>. Expires: 60. User-Agent: Sipean-AS 1.2.10. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: active;expires=60. Content-Length: 0. """ ###################### NOT BRIDGED, IN-DIALOG requests: ###################### """ # U 2014/11/12 19:46:18.448550 pub_ip_ua:54705 -> pub_ip_ep:5060 SUBSCRIBE sip:user1@192.168.51.168:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.243.82:5062;branch=z9hG4bK459444805. Route: <sip:pub_ip_ep;r2=on;lr;nat=yes>. Route: <sip:pub_ip_ep;transport=tcp;r2=on;lr;nat=yes>. From: "user2" <sip:user2@fakedomain.com>;tag=1902678123. To: <sip:user1@fakedomain.com>;tag=2518urgZJFet. Call-ID: 2104228355. CSeq: 3 SUBSCRIBE. Contact: <sip:user2@192.168.243.82:5062>. Accept: application/dialog-info+xml. Max-Forwards: 70. User-Agent: Tiptel IP 284 6.70.13.18 0015651b574f. Expires: 60. Event: dialog. Content-Length: 0. . # U 2014/11/12 19:46:18.449161 pub_ip_ep:5060 -> 192.168.51.168:5060 SUBSCRIBE sip:user1@192.168.51.168:5060 SIP/2.0. Via: SIP/2.0/UDP pub_ip_ep;branch=z9hG4bK7de2.a9aed4070706f93b25b4d2d3d7963a19.0. Via: SIP/2.0/UDP 192.168.243.82:5062;rport=54705;received=pub_ip_ua;branch=z9hG4bK459444805. From: "user2" <sip:user2@fakedomain.com>;tag=1902678123. To: <sip:user1@fakedomain.com>;tag=2518urgZJFet. Call-ID: 2104228355. CSeq: 3 SUBSCRIBE. Contact: <sip:user2@192.168.243.82:5062;alias=pub_ip_ua~54705~1>. Proxy-Authorization: Digest username="user2", realm="fakedomain.com", nonce="VGOr71RjqsMDoDyEqMnELO4fotFJLYhV", uri="sip:user1@192.168.51.168:5060", response="12fbfa50428b37bd55dbb4b51f8d5cba", algorithm=MD5. Accept: application/dialog-info+xml. Max-Forwards: 65. User-Agent: Tiptel IP 284 6.70.13.18 0015651b574f. Expires: 60. Event: dialog. Content-Length: 0. """

2 1

TLS/TCP sipping
by Petr Wozniak 11 Nov '14

11 Nov '14

Hello, Base on my testing OPTION pings can be sent only to UDP nated clients. So it is impossible with kamailio to send OPTION pings to TCP/TLS nated clients? I found doc for some version of nathelper module where the natping _tcp parameter can be set by modparam("nathelper", "natping_tcp", 1) Unfortunately, this parameter cannot be used in default nathelper module in kamailio 4.2.(doc is probably from OpenSIPS) I would like to send OPTION pings to TLS nated and individualy selected user agents. Does exist some possibility to do it with kamailio 4.2? If not by default does exist the possibility to compile and build kamailio with the nathelper module mentioned above (with natping_tcp parameter)? Or does exist some other solution? Thank you in advance Petr

1 0

Kamailio v4.2.1 Released
by Daniel-Constantin Mierla 11 Nov '14

11 Nov '14

Hello, Kamailio SIP Server v4.2.1 stable release is out. This is a maintenance release of the latest stable branch, 4.2, that includes fixes since release of v4.2.0. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.0. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.1. For more details about version 4.2.1 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/11/kamailio-v4-2-1-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

header manipulation in locally generated SIP OPTIONS requests
by Klaus Feichtinger 11 Nov '14

11 Nov '14

Hello, does kamailio offer any possibility for adding / replacing SIP headers in locally built SIP requests? In detail: SIP OPTIONS requests that are built by the DISPATCHER module for probing the configured targets. I have to extend these messages with "Accept" SIP header fields (which are marked with m* in RFC3261 - so they SHOULD be present), as the dispatcher target is requiring this information for answering these requests.... The event_route is principally accepting the append_hf() function and is adding the configured header fields. But the answer to that request (which is including these additional header fields) is confusing the dispatcher module. Dispatcher module is reacting with this ERROR message: DEBUG: dispatcher [dispatch.c:2406]: probing set #1, URI sip:10.10.10.10 DEBUG: dispatcher [dispatch.c:2345]: OPTIONS-Request was finished with code 500 (to , group 1) ERROR: dispatcher [dispatch.c:2358]: Setting the state failed (, group 1) When I comment the textops functions out again and use the original OPTIONS messages, dispatcher is happy and knows the states of the probed targets. Therefore, I´ll ask again: does kamailio offer any way for manipulating locally built SIP requests? kr Klaus

3 4

Releasing v4.2.1
by Daniel-Constantin Mierla 11 Nov '14

11 Nov '14

Hello, I am considering to release first minor version out of branch 4.2 as v4.2.1 next week, by Tuesday or Wednesday. If there is any comment regarding it, write to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com

1 1

ALIAS_DB replacing From and Contact fields
by Martin Schiøtz 11 Nov '14

11 Nov '14

Hi I'm using the ALIAS_DB module. username = 101 alias_username = 44444444 (fictive public phone number) 'alias_db_lookup("dbaliases")' is working fine for incoming calls from PSTN. Calling 44444444 R-URI is replaced with 101 SIP uri. But when 101 is calling to a PSTN number how do I replace the From and Contact fields with 44444444? Cheers, Martin

1 0

RTP end points
by Javier Ricke 10 Nov '14

10 Nov '14

i used asterisk, and install kamailio for traffic RTP can be send between end points, and do not have to go through RTP proxy, any work this form?, and client behind to NAT.. regards!

2 2

beginner's problem with sipping
by Petr Wozniak 10 Nov '14

10 Nov '14

Dear sr-users, I am beginner and apologize for bothering you with such a trivial problem. I would like to test sip UDP and TCP keepalive pinging. Kamailio and client are on the same network (192.168.1.44 and 192.168.1.20) that implies not behind NAT. So in kamailio.cfg I have set modparam("nathelper", "ping_nated_only", 0) My simple kamailio.cfg please you find enclosed. >From logfiles on client's side implies that neither UDP nor OPTIONS pings are sending from kamailio. If somebody would be so kind to help me I will be very glad. Thank you in advance Regards Petr

2 3

Counting xavp stack
by Alex Balashov 10 Nov '14

10 Nov '14

It would be great if there were some way to count the depth of a stack of XAVPs, e.g. $xavp(root=>num) = 1; $xavp(root[0]=>num) = 2; $xavp(root[0]=>num) = 3; ... $var(depth) = $(xavp(root=>num){avp.count}); Right now, the only way to do this that I know of is something like: $var(i) = 0; while(defined $xavp(root[$var(i)]=>num)) $var(i) = $var(i) + 1; $var(i) = $var(i) - 1; Is there a secret I don't know about? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 2

Setting up a IMS Server Platform using Kamailio 4.2.x code
by Austin Einter 10 Nov '14

10 Nov '14

Hi All Last few days we have tried to setup Kamailio for IMS server setup. We have not been successful. Now we are planning to use Kamailio 4.2.0 code base (Latest) to setup Kamailio IMS setup. I am looking for a link/document, that gives proper step by step instructions how to setup Kamailio as IMS servers (P-CSCF, I-CSCF and S-CSCF). Can somebody kindly provide the official link or any other working link, that I can follow to setup P-CSCF, I-CSCF ,S-CSCF and HSS. Many thanks Austin

2 1

Fwd: Extension and carrier same ip problem
by Martin Snelgoedkoop 10 Nov '14

10 Nov '14

> Hi, we use Kazoo which loads acl and dial plan out a DB. > We run in to smth that seems freeswitch default behavior so that's why I write here. > > We have a carrier that needs to register a extension to our platform as well. > They use the same ip. And that creates a problem as when I place a call with the device using the extension it gets identified as carrier and the route is then not correct. > > The same happens the other way arround, if a call comes in via the carrier the ip is recognized as an extension ip and then it won't use the carrier route. > > I was wondering if I could use some flags or a way of changing the ip used or doing some magic to make it work. I would prefer not to use a proxy. > > The setup is kamailio>freeswitch > > > Met vriendelijke groet, Martin > > Team snelgoedkoop

3 2

Kamilio verify online servers
by Marino Mileti 10 Nov '14

10 Nov '14

Hi guys, I've an architecture with multiple Kamailio. I would like to implement some check that allow me to verify if a server is online or not so for example i can use msilo to save chat message and relay it when the server come's up. Is it possible to use UAC module to send a REGISTER message from one server to another so i can "dump" all message in very simple way? Is there any example ? Many thanks ----- Marino Mileti -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamilio-verify-online-servers-tp132… Sent from the Users mailing list archive at Nabble.com.

2 1

htable locking
by Alex Balashov 10 Nov '14

10 Nov '14

The documentation for the htable module gives no guidance on whether sht_lock()/unlock() should be used in all situations when writing or reading from the htable. Should it be? I was under the impression (not supported by anything said in the documentation) that read and write access to the htable is fundamentally thread-safe, i.e. it is safe to both read and write to the htable even if the potential exists for another worker process to do the same concurrently. Is that not the case? If not, should there not be strong admonitions to use sht_lock/sht_unlock around concurrent operations--and it sounds like nearly all useful applications of the htable are essentially concurrent? Thanks! -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

3 6

Generate REFER msg
by Agiftel 10 Nov '14

10 Nov '14

Hi all, i need some advice on how to solve a problem. I've a scenario like following: [SIP PBX] <---->CUCM (Call Manager v 10.5 ) User A: belong to CUCM User B: belong to SIP PBX User C: belong to SIP PBX User A calls User B. After some conversation, User A wants to transfer call to User C. CUCM transfer the call with re-INVITE; in this case on SIP PBX 2 channel SIP are used. CUCM does not support transfer, on Trunk, using REFER method. My idea is using Kamailio beetwen SIP-PBX and CUCM and handles call generating REFER msg in some way. Is this idea a good one? Or there is a better way? Regards -- View this message in context: http://sip-router.1086192.n5.nabble.com/Generate-REFER-msg-tp132317.html Sent from the Users mailing list archive at Nabble.com.

2 1

rr problem: sips instead of sip
by Jon Bonilla (Manwe) 07 Nov '14

07 Nov '14

Hi all I'm experiencing a problem while using a mobile client on a NGCP mr3.5.1 (based on kamailio 4.1.6) The problem is as follows: A custom mobile client B blink for linux The path of a call from B to A would be this one: B --> Kamailio LB --> Kamailio proxy --> Sems --> Kamailio LB --> A Kamailio LB is a stateless proxy. Kamailio proxy is a stateful proxy and registrar and sems acts as B2BUA. Kamailio LB listens tcp,udp and tls to the outside and speaks udp to the inside. Kamailio proxy and sems only listen udp localhost (ports 5062 ans 5080). My problem comes when the ACK from B to A (to the INVITE-200) doesn't leave the Kamailio LB to the proxy. The problem is that the kamailio proxy added itself as sips in the record-route header: U 2014/11/06 19:22:20.447717 127.0.0.1:5062 -> 127.0.0.1:5080 INVITE sips:test1@10.0.1.5:41956;rinstance=145658D8;transport=tcp SIP/2.0' Record-Route: <sips:127.0.0.1:5062;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;did=3e6.48e1;mpd=ii;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=aHRBd2tQQ3VhVkZGamdrN2lhanhodEF3aw-->' Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;nat=yes;ngcplb=yes;socket=udp:XX.XX.XX.XX:5060>' Record-Route: <sip:XX.XX.XX.XX;r2=on;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;nat=yes;ngcplb=yes;socket=udp:XX.XX.XX.XX:5060>' Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bKc36a.b42be7e4a4db27ac699a52fda7fb5fe0.0' Route: <sip:lb@127.0.0.1;lr;received=sip:YY.YY.YY.YY:41956%3Btransport%3Dtls;socket=sip:XX.XX.XX.XX:5061>' Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc36a.a58b9955ce93726b17c0240a555011f2.0' Via: SIP/2.0/UDP 10.10.10.4:47879;received=ZZ.ZZ.ZZ.ZZ;rport=47879;branch=z9hG4bKPj6UmIovzyxDRCgcuz-VoXHEyJHRpA712x' When the ACK arrives to the LB from B, it can't relay it to the Kamailio proxy: NOTICE: <script>: New request on lb - M=ACK R=sip:ngcp-lb@XX.XX.XX.XX;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31 NOTICE: <script>: Relaying request, du='sips:127.0.0.1:5062;lr;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;did=3e6.48e1;mpd=ii;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=aHRBd2tQQ3VhVkZGamdrN2lhanhodEF3aw--', fs='udp:127.0.0.1:5060' - R=sip:127.0.0.1:5080;prxroute=1 WARNING: <core> [forward.c:264]: get_send_socket2(): WARNING: get_send_socket: protocol/port mismatch (forced udp:127.0.0.1:5060, to tls:127.0.0.1:5062) I think the problem might be in the way the client registers? This is the registration info where it uses sips in the contact: contact: sips:test1@10.0.1.5:41956;rinstance=145658D8;transport=tcp If I try the same calling blink, which has the following contact in the registration it works without issues as the proxy doesn't insert sips in the Record-Route header: contact: sip:91807432@10.10.10.4:57160;transport=tls Any thoughts? cheers, Jon

3 2

Kamailio v4.0.7 Released
by Daniel-Constantin Mierla 07 Nov '14

07 Nov '14

Hello, Kamailio SIP Server v4.0.7 stable release is out. This is a maintenance release of the 3rd old stable branch, 4.0, that includes fixes since release of v4.0.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.0.x. Deployments running previous v4.0.x versions are strongly recommended to be upgraded to v4.0.7, or even better, to 4.1.x or 4.2.x, as the focus from now on will be on maintenance of these release series. For more details about version 4.0.7 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/11/kamailio-v4-0-7-released/ Note: the latest stable branch is 4.2, at this moment with its latest release v4.2.0. See more details about it at: * http://www.kamailio.org/w/2014/10/kamailio-v4-2-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Releasing v4.0.7
by Daniel-Constantin Mierla 07 Nov '14

07 Nov '14

Hello, I am planning to release v4.0.7 based on latest source code in branch 4.0. This branch is going out of official maintenance, the current latest stable branches being 4.2 and 4.1. If you have any remarks regarding it, write back to mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com

1 2

cdp.so segment fault
by Tom Secula 06 Nov '14

06 Nov '14

I am just wondering if anyone has seen this type of segfault on 4.2 ? I'm trying to run kamailio as an scscf on centos 6.6 ? Nov 6 14:17:20 scscf kernel: kamailio[6246]: segfault at 0 ip 00f3f425 sp bfa0e9f4 error 6 in cdp.so[f08000+96000]

1 0

Re: [SR-Users] RPC uac.reg_info hangs second time
by Dan Christian Bogos 06 Nov '14

06 Nov '14

Hey Daniel, Thanks for your fast reply! I am using the 4.2.0 out of debian packages (not nightly). Will shift on nightly and retest. Not a big deal regarding 150 seconds, maybe it should be a good reason for async reloads (reply with the delay number of seconds when reload will happen). Reloads can be always implemented outside of kamailio but since everybody would need it I assume a more centralized approach would be useful. Anyway, thanks again! DanB On 24.10.2014 12:00, sr-users-request(a)lists.sip-router.org wrote: > Hello, > > are you using branch 4.2 or some tgz/package from 4.2.0 release? There > was a fix related to such issue, so 4.2 branch must be used (or debian > nightly builds for 4.2). > > I will have to check the code how flexible that is, but afaik, the limit > on reload interval was thought to avoid sending again registration > requests while retransmissions from previous one can occur. > > Cheers, > Danel

2 7

Kamailio v4.1.7 Released
by Daniel-Constantin Mierla 06 Nov '14

06 Nov '14

Hello, Kamailio SIP Server v4.1.7 stable release is out. This is a maintenance release of the previous stable branch, 4.1, that includes fixes since release of v4.1.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.7. For more details about version 4.1.7 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/11/kamailio-v4-1-7-released/ Note: the latest stable branch is 4.2, at this moment with its latest release v4.2.0. See more details about it at: * http://www.kamailio.org/w/2014/10/kamailio-v4-2-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

how to check if reply is retransmission?
by Juha Heinanen 06 Nov '14

06 Nov '14

is there a means to check if reply to t_relay()ed request is retransmission? there is t_is_retr_async_reply(), but that does not apply to regular transactions. i would like to avoid double calling of rtpengine_answer(). -- juha

4 5

AT&T and UAC cancel after 200 OK
by Yuriy Gorlichenko 06 Nov '14

06 Nov '14

Hello. I use UAC module for trunks and have trouble when call to mobile endpoint of AT&T provider. When endpoint pickup call ans 200 OK reply come into kamailio, kamailio send CANCEL. so at the endpoint I see than call is dropped and at kamailio client I hear voicemail speech form AT&T endpoint. Provider say that his trace is OK... My sip trace bellow: IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1129 E.......@.l. ...6........q..INVITE sip:98765432100@phone.myprovider.com SIP/2.0 Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Contact:<sip:12345678@sip.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Mon, 03 Nov 2014 20:21:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 314 v=0 o=root 1363870848 1363870848 IN IP4 68.34.22.11 s=Asterisk PBX 12.5.0 c=IN IP4 68.34.22.11 t=0 0 m=audio 30030 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30031 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 651 E.......*.Z.6... .........}.SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0;received=68.34.22.11 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com >;tag=04e2a294d0728b89107e081a19babab4.43a7 Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="phone.myprovider.com", nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", qop="auth" Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 410 E.......@.o. ...6.........j%ACK sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com >;tag=04e2a294d0728b89107e081a19babab4.43a7 Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 ACK Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1402 E.......@.k. ...6..........`INVITE sip:98765432100@phone.myprovider.com SIP/2.0 Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Contact:<sip:12345678@sip.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Mon, 03 Nov 2014 20:21:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 314 Proxy-Authorization: Digest username="12345678", realm="phone.myprovider.com", nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", uri=" sip:98765432100@phone.myprovider.com", qop=auth, nc=00000001, cnonce="2107612791", response="51db231dedb4a74396447729a192a8d3", algorithm=MD5 v=0 o=root 1363870848 1363870848 IN IP4 68.34.22.11 s=Asterisk PBX 12.5.0 c=IN IP4 68.34.22.11 t=0 0 m=audio 30030 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30031 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1402 E.......@.k. ...6.........._INVITE sip:98765432100@phone.myprovider.com SIP/2.0 Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Contact:<sip:12345678@sip.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Mon, 03 Nov 2014 20:21:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 314 Proxy-Authorization: Digest username="12345678", realm="phone.myprovider.com", nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", uri=" sip:98765432100@phone.myprovider.com", qop=auth, nc=00000001, cnonce="2107612791", response="51db231dedb4a74396447729a192a8d3", algorithm=MD5 v=0 o=root 1363870848 1363870848 IN IP4 68.34.22.11 s=Asterisk PBX 12.5.0 c=IN IP4 68.34.22.11 t=0 0 m=audio 30030 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30031 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 529 E..-....*.[.6... .........xiSIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102;rport=1024;received=68.34.22.11 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 529 E..-....*.[.6... .........xkSIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102;rport=1024;received=68.34.22.11 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 998 E.......*.Y16... .........^.SIP/2.0 180 Ringing Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 986 E.......*.Y<6... ...........SIP/2.0 180 Ringing Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=UrBQrZp9vSm3p Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE Contact: <sip:98765432100@173.203.60.50:5060;transport=udp> User-Agent: myprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1294 E..*....*.X.6... .........:.SIP/2.0 183 Session Progress Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1282 E.......*.X.6... ........ ..SIP/2.0 183 Session Progress Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=UrBQrZp9vSm3p Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE Contact: <sip:98765432100@173.203.60.50:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415020637 1415020638 IN IP4 173.203.60.50 s=FreeSWITCH c=IN IP4 173.203.60.50 t=0 0 m=audio 25430 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W.6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 412 E.......@.o. ...6.........j.CANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 CANCEL Content-Length: 0 Reason: SIP;cause=200 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6...........ACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.0ec383b6dc8fbb1861dfe644f92ee9ca.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK010bb317;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W~6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6..........jACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.6a1e71bf5fcd88f137733279fa31770c.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK1f95579c;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406 E.......@.o. ...6.........?xCANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 CANCEL Content-Length: 0 Reason: SIP;cause=200 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 246 E.......*.\.6... ...........OPTIONS sip:68.34.22.11:1024 SIP/2.0 Via: SIP/2.0/UDP 10.202.129.254:5060;branch=0 From: sip:keepalive@myprovider.com;tag=29ebf5d To: sip:68.34.22.11:1024 Call-ID: 6360a4bb-53bebf7-5686ca6(a)10.202.129.254 CSeq: 1 OPTIONS Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 330 E..f....@.p. ...6........R.<SIP/2.0 405 Method not allowed Via: SIP/2.0/UDP 10.202.129.254:5060;branch=0;rport=5060;received=34.43.4.23 From: sip:keepalive@myprovider.com;tag=29ebf5d To: sip:68.34.22.11:1024;tag=c6894543322e2a4942d921e4298ce904.1ec7 Call-ID: 6360a4bb-53bebf7-5686ca6(a)10.202.129.254 CSeq: 1 OPTIONS Server: MS Lync Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W|6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6..........eACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.47488c128d3518b837410e363acc963e.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK5e2d222b;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406 E.......@.o. ...6.........?xCANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 CANCEL Content-Length: 0 Reason: SIP;cause=200 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W{6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6..........;ACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.00885667802b30315ce7f8b5db31e2bd.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK6081ac89;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406 E.......@.o. ...6.........?xCANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 CANCEL Content-Length: 0 Reason: SIP;cause=200

1 0

how to use mqueue.
by 张顺通 06 Nov '14

06 Nov '14

whle (mq_fetch("QUEUE")) { if (yes) { do something } else { mq_add("QUEUE"); add back to queue. } } can mq_add use here? I tried it, it do not work. and How to determine the end?

3 6

Releasing v4.1.7
by Daniel-Constantin Mierla 06 Nov '14

06 Nov '14

Hello, I plan to release a new version (4.1.7) of the previous stable branch 4.1. Most probably on Thursday this week. If anyone has remarks/issues about it, write back to mailing lists to discuss. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com

1 1

benchmark module
by bob sacamano 05 Nov '14

05 Nov '14

hello, i was wondering where does this module write the logs? Nothing shows up in my syslog. If i replace the benchmark parts in the route with xlog for example xlog( "L_INFO", "$Ts " ); this shows up fine in syslog heres what u have in my config file: modparam("benchmark", "enable", 1) modparam("benchmark", "granularity", 100) modparam("benchmark", "loglevel", 3) if(is_method("REGISTER")) { bm_start_timer("test"); save("location"); bm_log_timer("test"); exit; } I have been searching but cant find any info on it

2 5

Kamailio with media proxy and two network interfaces
by Julien 05 Nov '14

05 Nov '14

Hello, I have a problem with Kamailio and a setup with two network interfaces (eth0 and eth1) Devices on eth0 CANNOT see devices on eth1 and vice versa, there is no route between both interfaces. Only Kamailio and the media proxy can access both. +--------------------------+ | kamailio + rtpproxy | eth0 | + openvpn | eth1 +---------+ +-----------+ | | | | | +--------------------------+ | +-+----+ +----+-+ | A | | B | | | | | +------+ +------+ A user connected at eth0 shall be able to get the presence status of users on eth1 and users shall be able to call each other. What I did so far: I added listen statements for both interfaces to the Kamailio config file: -listen=udp:eth0_IP:5060 -listen=udp:eth1_IP:5060 What's not working: When I try to make a call from user in A domain to another user in B domain, the call established correctly, but no stream running. On B device I see an incoming call from 'sip:user@A_Domain'. B user doesn't have route for A Domain, so that doesn't work. Is what I want technically possible with Kamailio. If yes, what do I have to change in my config? Do you have any suggestions? Thanks in advance. -- Julien Barreau

3 3

how you make Kamailio with mariadb
by ANTHONY HESNAUX 05 Nov '14

05 Nov '14

hello Daniel I want to install Kamailio with mariadb on CENTOS 7 I already install Kamailio on CENTOS 6.5 without error but on CENTOS 7 with mariadb, What modules do you include for make cfg: make include_modules="db_mysql" cfg thank you best regard Anthony HESNAUX

3 8

UAC error at reply : t_should_relay_response:
by Yuriy Gorlichenko 05 Nov '14

05 Nov '14

ERROR: t_should_relay_response: status rewrite by UAS: stored: 491, received: 200 Hello. I use UAC module for trunks form 4.2 master branch. I have an error when endpoit at the trunk picked up. OK reply never replied to caller because I see next error: ERROR: t_should_relay_response: status rewrite by UAS: stored: 491, received: 200 So as I see before reply 200 sended to kamailio (from callee) I see packet with 491 SIP code. It is strange because at the another server (same configuration) OK successfully sended to caller and session is Ok...

1 0

rare crash , race condition between 183 and 408 ?
by Dragos Oancea 05 Nov '14

05 Nov '14

Hi all & hi Daniel , We had a crash with version 4.0.5 today . I looks like it happens due to a race condition between a provisional response (183) and the generation of a 408 . We send the INVITE, get a `100 Trying` from upstream , then we get a `183 Session Progress` from upstream , then after exactly 1 minute , we get another `183 Session Progress` from upstream. At this moment kamailio crashed. Some tm params we have : modparam("tm", "fr_timer", 5000) modparam("tm", "fr_inv_timer", 60000) <- our final reply timer is 1 minute . modparam("tm", "restart_fr_on_each_reply", 1) GDB output and logs here: kamailio 4.0.5 crash - in fake_req() - race cond - Pastebin.com kamailio 4.0.5 crash - in fake_req() - race cond - Paste... (gdb) bt #0 0x00007f4d93129a32 in memcpy () from /lib64/libc.so.6 View on pastebin.com Preview by Yahoo In fake_req() from t_reply.c , the pointer shmem_msg->new_uri.s which is passed to memcpy becomes invalid. memcpy( faked_req->new_uri.s, shmem_msg->new_uri.s,faked_req->new_uri.len); (gdb) p shmem_msg->new_uri $3 = {s = 0x0, len = 0} If this bug is not supposed to be fixed in 4.2.0 , we can help with debugging . Regards, Dragos

2 1

AliasDB module & Forking
by Marino Mileti 05 Nov '14

05 Nov '14

Hi guys, i've implemented a parallel forking using aliasdb module. Everything is working good if the first client (first row in dbaliases table) in online. When I try to make a call to a specific number three clients rings together. If the first client isn't online no call is forwarded to other clients :( I know that with alias_db_lookup() the first record from DB is the new R-URI and the others are added like branches. But if the R-URI in unreachable how can try to call the other branches? This is location script.. if(alias_db_lookup("dbaliases")){ $sht(a=>$ci) = $sel(contact.uri); } $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: xlog("L_INFO","No contact found\n"); exit; case -3: send_reply("404", "Erorr"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } Any suggest? My script still log "No contanct found" until timeout :( Many thanks ----- Marino Mileti -- View this message in context: http://sip-router.1086192.n5.nabble.com/AliasDB-module-Forking-tp132128.html Sent from the Users mailing list archive at Nabble.com.

2 3

DTMF Tones question
by Anthony Martin 05 Nov '14

05 Nov '14

HI, Please, excuse me for my bad English Just a question, kamailio does support DTMF Tones? i have a big while searching for documentation about that but without good results. I want to make a server side call transfer method whit DTMF Tones... Thanks so much for your good work Greetings from Venezuela

3 2

DBURL
by shmick＠riseup.net 05 Nov '14

05 Nov '14

[i have no idea about email subject nor anything else for this except prior i configured rtpproxy to work with kamailio on a unix socket and updated tls.cfg to use my cert/key] running kamailio from deb http://deb.kamailio.org/kamailio jessie main $ sudo service kamailio restart [FAIL] Not starting Kamailio SIP Server: invalid configuration file! ... failed! failed! [....] 0(15228) INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... 0(15228) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 353, column 30-34: syntax error 0(15228) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 353, column 30-34: Invalid arguments 0(15228) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 353, column 35: ERROR: bad config file (3 errors) 0(15228) WARNING: <core> [ppcfg.c:219]: pp_ifdef_level_check(): different numbe[FAILpreprocessor directives: N(#!IF[N]DEF) - N(#!ENDIF) = 1 ... failed! failed! /* 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*) <--------not happy with DBURL here modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif

2 3

dnssec module
by shmick＠riseup.net 05 Nov '14

05 Nov '14

running kamailio from deb http://deb.kamailio.org/kamailio jessie main when i add the dnssec module to /etc/kamailio/kamailio.cfg kamailiop fails to start and gives: [FAIL] Not starting Kamailio SIP Server: invalid configuration file! ... failed! failed! [....] 0(8839) ERROR: <core> [sr_module.c:597]: load_module(): could not open module </usr/lib/x86_64-linux-gnu/kamailio/modules/dnssec.so>: /usr/lib/x86_64-linux-gnu/libval-threads.so.14: undefined symbol: SSL_get_SSL_CTX 0(8839) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /etc/kamailio/kam[FAIL.cfg, line 239, column 12-22: failed to load module ERROR: bad config file (1 errors) ... failed! failed! $ sudo ls -la /usr/lib/x86_64-linux-gnu/kamailio/modules/dnssec.so -rw-r--r-- 1 root root 22912 Oct 17 09:04 /usr/lib/x86_64-linux-gnu/kamailio/modules/dnssec.so any help appreciated thanks

2 4

REGISTRAR location replication (pua_reginfo)
by lejarreta.e＠sarenet.es 04 Nov '14

04 Nov '14

Good morning. We've got a problem with location memory and database replication between "n" kamailios as REGISTRAR servers. 1.- WHAT WE WANNA DO: - To have "n" Kamailios with their DB and memory "user location" replicated. - Each Kamailio owns its own DB.(It could be also with a common DB). - Each kamailio has a PUA_REGINFO, PRESENCE, and REGISTRAR roles. - USRLOC Module with db_mode = 2 (Write-Back scheme). 2.- WHAT WE DON'T WANNA DO: - USRLOC Module with db_mode != 2 (Write-Back scheme). 3.- BACKGROUND: - RFC3265 - RFC3680 - RFC3903 - I attach a small architecture graph. 4.- PROBLEMS: 4.1 Deadlock: Each REGISTER involves a NOTIFY to the other side. Each NOTIFY involves another NOTIFY to the other side. There should be a mechanism to avoid PUBLISH reginfo_handle_notify() function calls but to PUBLISH save() calls on pua_reginfo module. Any idea would be appreciated (also to workaround our problem). 4.2 PUA_REGINFO module notify.c Just FYI. We think It doesn't comply to RFC3680. It should observe "unkown-param" object to aid "Contact" header params such as "Path", "Received", "User_agent" and so on. We've changed code but for a fast startup we've also avoided "unkown-param" and now we're able to see replicated "Path", "Received ... params on each DB. Thanks and best regards. Eduardo Lejarreta.

2 1

how to configure multi-proxy mode for kamailo 4.2.x
by andrew 04 Nov '14

04 Nov '14

hi By default, kamailio.cfg is used for one proxy. How to set up kamailio to support multi-proxy, so that serveral kamaiio servers work together, and each one is responsible for sip clients in some specific area? B.R. Andrew

2 1

"Help with dialog $dlg_var(cseq_diff)"
by Yuriy Gorlichenko 03 Nov '14

03 Nov '14

Hello. I need to increment CSeq value for INVITE with Auth params when use UAC_AUTH for outgoing calls to provider. Kamailio 4.2 may increment this using dialog module http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-7-increment-cseq-for… Now I experements with this and var $dlg_var(cseq_diff) and at transaction way int have NULL value. I can not understand why My config is: # ----- dialog params modparam("dialog", "db_url",DBURL) modparam("dialog", "db_mode", 1) modparam("dialog","table_name","dialog") modparam("dialog", "dlg_flag", 4) modparam("dialog", "initial_cbs_inscript", 1) modparam("dialog", "profiles_with_value", "caller") modparam("dialog", "default_timeout", 60) modparam("dialog", "track_cseq_updates", 1) route config is at request route if(is_method("INVITE") && !has_totag()){ $dlg_ctx(timeout_route) = "DIALOG_END"; $dlg_ctx(timeout_bye) = 1; dlg_manage(); xlog("L_INFO","Dialog manage is {$ct}\n"); } } t_relay(); next I handle failure reply because 407 reply is recieved failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_check_status("401|407")){ xlog("L_INFO", "Reply from provider on failure: $tU"); xlog("L_ERR","401/407 - Unauthorized. ($ci .) ($rm) from ($fu) (IP:$si:$sp) to ($Ri:$Rp). Must be authorized with digest Auth."); avp_print(); xlog("L_INFO", "CSeq diff: $dlg_var(cseq_diff)"); uac_auth(); xlog("L_INFO", "UAC_AUTH(): $tU"); append_branch(); t_relay(); } if (t_is_canceled()) { exit; } } As you may see i logging $dlg_var(cseq_diff) value and now it NULL. So I can not understand why? What wrong I do? Thanks for advice.

2 3

Problems using htable iterator
by Glenn Larsen 03 Nov '14

03 Nov '14

Hi, I have a hard time using the htable iterator... I have this simple setup: modparam("htable", "htable", "h1=>size=4") ... sht_iterator_start("i1", "h1"); while (sht_iterator_next("i1")) { xlog("L_ERR", " SS: $shtitkey(i1) and $shtitval(i1)\n"); } sht_iterator_end("i1"); ... Now to put some simple data, I use: # sudo kamcmd htable.sets h1 test 123 # sudo kamcmd htable.dump h1 { entry: 8 size: 1 slot: { item: { name: test value: 123 } } } When the route is "executed", I get this in my log file: Nov 3 15:16:47 beast /sbin/kamailio[24865]: ERROR: SS: <null> and <null> The $shtitkey(i1) and $shtitval(i1) is always null. Is it a bug? Or did I manage to do something wrong? -- Best regards, Glenn O. Larsen -- Best regards, *Glenn O. Larsen* *CTO* *SimCom AS* Professor Birkelands vei 24c N-1081 Oslo Direct: +47 21 09 34 12 Mobile: +47 997 00 000 Online profile: http://simcom.no/glenn.larsen *We simplify your communication ...*

2 1

rtpengine does not send RTP packets out
by Juan Perez 03 Nov '14

03 Nov '14

Hi, I have kamilio-4.2 and rtpengine running on the same machine.I have this scenario: softphone --> Kamailio with Rtpengine --> AsteriskThe softphone initiates the call, it is sent to the Asterisk. I can see the SDPs being re-written with the new IP/Ports provided by rtpengine: Invite from Kamailio to Asterisk200 Ok from Kamailio to Softphone However, I take a signaling/media capture on the server where the kamailio/rtpengine are running and see the RTP coming from both endpoints (softphone and asterisk) to the correct ports but there is no packets coming out from the proxy to either direction. I see these 2 lines on the rtpengine log and make me think that something prevents the rtpengine to stream out to the 2 endpoints: Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Write error on RTP socket Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Final packet stats: this is the log from the rtpengine: Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: Got valid command from 127.0.0.1:46686: offer - { "sdp": "v=0#015#012o=- 314474204173996793 6663595733338147866 IN IP4 99.153.161.148#015#012s=-#015#012c=IN IP4 99.153.161.148#015#012t=0 0#015#012m=audio 16386 RTP/AVP 114 0 8 101#015#012a=rtpmap:114 ilbc/8000#015#012a=fmtp:114 mode=30#015#012a=rtpmap:0 pcmu/8000#015#012a=rtpmap:8 pcma/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012", "direction": [ "pub", "pub" ], "replace": [ "origin", "session-connection" ], "call-id": "0866b358-dc9c-1232-1399-3767db69b8dd", "received-from": [ "IP4", " ... Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: ... 99.153.161.148" ], "from-tag": "73rH43p60NeKm", "command": "offer" } Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Creating new call Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Opened ports 35022..35023 for media relay Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Opened ports 35038..35039 for media relay Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Returning to SIP proxy: d3:sdp529:v=0#015#012o=- 314474204173996793 6663595733338147866 IN IP4 54.173.60.13#015#012s=-#015#012c=IN IP4 54.173.60.13#015#012t=0 0#015#012a=ice-lite#015#012m=audio 35022 RTP/AVP 114 0 8 101#015#012a=rtpmap:114 ilbc/8000#015#012a=fmtp:114 mode=30#015#012a=rtpmap:0 pcmu/8000#015#012a=rtpmap:8 pcma/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=rtcp:35023#015#012a=ice-ufrag:oPm3I7OE#015#012a=ice-pwd:WjZnh7dnhfdkfoXDoqDbnCLLr7N7#015#012a=candidate:NZ1IXXQHYxOora9j 1 UDP 2130706431 54.173.60.13 35022 typ host#015#012a=candidate: ... Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] ... NZ1IXXQHYxOora9j 2 UDP 2130706430 54.173.60.13 35023 typ host#015#0126:result2:oke Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: Got valid command from 127.0.0.1:46686: answer - { "sdp": "v=0#015#012o=root 439969655 439969655 IN IP4 205.10.248.62#015#012s=Asterisk PBX 1.8.10.0#015#012c=IN IP4 205.10.248.62#015#012t=0 0#015#012m=audio 11142 RTP/AVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=silenceSupp:off - - - -#015#012a=ptime:20#015#012a=sendrecv#015#012", "direction": [ "pub", "pub" ], "replace": [ "origin", "session-connection" ], "call-id": "0866b358-dc9c-1232-1399-3767db69b8dd", "received-from": [ "IP4", "205.10.248.62" ], "fr ... Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: ... om-tag": "73rH43p60NeKm", "to-tag": "as57e7b8e3", "command": "answer" } Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Returning to SIP proxy: d3:sdp500:v=0#015#012o=root 439969655 439969655 IN IP4 54.173.60.13#015#012s=Asterisk PBX 1.8.10.0#015#012c=IN IP4 54.173.60.13#015#012t=0 0#015#012a=ice-lite#015#012m=audio 35038 RTP/AVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=silenceSupp:off - - - -#015#012a=ptime:20#015#012a=sendrecv#015#012a=rtcp:35039#015#012a=ice-ufrag:cBL9SIw9#015#012a=ice-pwd:QHw7VHp83nMsLpD8ZoSqtDxGchN4#015#012a=candidate:NZ1IXXQHYxOora9j 1 UDP 2130706431 54.173.60.13 35038 typ host#015#012a=candidate:NZ1IXXQHYxOora9j 2 UDP 213070 ... Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] ... 6430 54.173.60.13 35039 typ host#015#0126:result2:oke Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Write error on RTP socket Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Final packet stats: Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] --- Tag 'as57e7b8e3', created 0:00 ago, in dialogue with '73rH43p60NeKm' Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] ------ Media #1, port 35022 <> 205.10.248.62:11142, 0 p, 0 b, 0 e Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] ------ Media #1, port 35023 <> 205.10.248.62:11143 (RTCP), 0 p, 0 b, 0 e Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] --- Tag '73rH43p60NeKm', created 0:00 ago, in dialogue with 'as57e7b8e3' Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] ------ Media #1, port 35038 <> 99.153.161.148:16386, 0 p, 0 b, 1 e Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] ------ Media #1, port 35039 <> 99.153.161.148:16387 (RTCP), 0 p, 0 b, 0 e Nov 1 18:59:36 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Got valid command from 127.0.0.1:46686: delete - { "call-id": "0866b358-dc9c-1232-1399-3767db69b8dd", "received-from": [ "IP4", "205.10.248.62" ], "from-tag": "as57e7b8e3", "command": "delete" } Nov 1 18:59:36 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] [0866b358-dc9c-1232-1399-3767db69b8dd] Call-ID to delete not found Nov 1 18:59:36 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Returning to SIP proxy: d7:warning38:Call-ID not found or tags didn't match6:result2:oke this is the log from kamailio (showing only the lines that include the word rtpengine): Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: INFO: <script>: mylog: TO_FS: Starting rtpengine session. Invite Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: exec: *** cfgtrace:request_route=[TO_FS] c=[/usr/local/kamailio-install-dir//etc/kamailio//kamailio.cfg] l=441 a=25 n=rtpengine_offer Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine_funcs.c:148]: check_content_type(): type <application/sdp> found valid Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine.c:1448]: rtpp_function_call(): proxy reply: d3:sdp529:v=0#015#012o=- 314474204173996793 6663595733338147866 IN IP4 54.173.60.13#015#012s=-#015#012c=IN IP4 54.173.60.13#015#012t=0 0#015#012a=ice-lite#015#012m=audio 35022 RTP/AVP 114 0 8 101#015#012a=rtpmap:114 ilbc/8000#015#012a=fmtp:114 mode=30#015#012a=rtpmap:0 pcmu/8000#015#012a=rtpmap:8 pcma/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=rtcp:35023#015#012a=ice-ufrag:oPm3I7OE#015#012a=ice-pwd:WjZnh7dnhfdkfoXDoqDbnCLLr7N7#015#012a=candidate:NZ1IXXQHYxOora9j 1 UDP 2130706431 54.173.60.13 35022 typ host#015#012a=candidate:NZ1IXXQHYxOora9j 2 UDP 2130706430 54.173.60.13 35023 typ host#015#0126:result2:oke Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: INFO: <script>: mylog: Starting rtpengine session. Answer Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: exec: *** cfgtrace:onreply_route=[TO_FS] c=[/usr/local/kamailio-install-dir//etc/kamailio//kamailio.cfg] l=476 a=25 n=rtpengine_answer Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine_funcs.c:148]: check_content_type(): type <application/sdp> found valid Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine.c:1448]: rtpp_function_call(): proxy reply: d3:sdp500:v=0#015#012o=root 439969655 439969655 IN IP4 54.173.60.13#015#012s=Asterisk PBX 1.8.10.0#015#012c=IN IP4 54.173.60.13#015#012t=0 0#015#012a=ice-lite#015#012m=audio 35038 RTP/AVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=silenceSupp:off - - - -#015#012a=ptime:20#015#012a=sendrecv#015#012a=rtcp:35039#015#012a=ice-ufrag:cBL9SIw9#015#012a=ice-pwd:QHw7VHp83nMsLpD8ZoSqtDxGchN4#015#012a=candidate:NZ1IXXQHYxOora9j 1 UDP 2130706431 54.173.60.13 35038 typ host#015#012a=candidate:NZ1IXXQHYxOora9j 2 UDP 2130706430 54.173.60.13 35039 typ host#015#0126:result2:oke Nov 1 18:59:36 ip-10-0-2-68 ./kamailio[28001]: exec: *** cfgtrace:request_route=[LOOSE_ROUTE] c=[/usr/local/kamailio-install-dir//etc/kamailio//kamailio.cfg] l=488 a=24 n=rtpengine_delete Nov 1 18:59:36 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine.c:1448]: rtpp_function_call(): proxy reply: d7:warning38:Call-ID not found or tags didn't match6:result2:oke

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