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How can i change kamailio SIP port form 5060 to any other
by Gaurav Bmotra 17 Apr '19

17 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i want to change kamailio SIP port 5060 to other port for security purpose plz help me -- *Regards:* Gaurav Kumar

2 3

TCP connection lifetime
by Aymeric Moizard 17 Apr '19

17 Apr '19

Hi Again, Here is an issue with TCP connection being kept for more: Yesterday, I have discovered that a User-Agent (<Avaya IP Phone 1120E (SIP1120e.04.04.30.00)> tried to register a lot. It was sending REGISTER over new established TCP socket *every 2 seconds*. All the REGISTER was rejected with 401. (may be the device was misconfigured? or not receiving any of my answer? I can't tell) NOTE: You can see the expires header was very large: 86400, ie: 24 hours... I was checking the TCP/TLS connections on my server and discovered more than 1000 TCP established connection to that user/ip, and thus, I have tried to understand what happened. Checking the logs, I received 4855 REGISTER from this device from "Mar 25 03:47:09" to "Mar 25 07:56:13" which is a rate of approx one new TCP connection every 2.5 seconds... Today, I decided to check it again around 11am. jack@sip:~$ sudo kamctl stats tcp { "jsonrpc": "2.0", "result": [ "tcp:con_reset = 1857", "tcp:con_timeout = 35927", "tcp:connect_failed = 25", "tcp:connect_success = 2", "tcp:current_opened_connections = 2291", "tcp:current_write_queue_size = 0", "tcp:established = 80778", "tcp:local_reject = 0", "tcp:passive_open = 80776", "tcp:send_timeout = 2", "tcp:sendq_full = 0" ], "id": 7305 } There was still A LOT of established connections. And the connections have been established more than 24 hours ago. At 11H16: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 1161 At 11H22: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 1018 At 11H35: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 655 At 13H $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 0 So the established connections are all gone now. Between 11h16 and 11H35, I was seeing the server regularly sending [FIN, ACK] over each TCP established connection, with retransmissions for all of them. (no incoming trafic) I do not have numbers/capture/stats, but I think that kamailio was already closing some connection yesterday. I don't know when kamailio started to try closing those connections. I'm now back with this status: At 13pm: jack@sip:~$ sudo kamctl stats tcp { "jsonrpc": "2.0", "result": [ "tcp:con_reset = 1896", "tcp:con_timeout = 38042", "tcp:connect_failed = 26", "tcp:connect_success = 2", "tcp:current_opened_connections = 939", "tcp:current_write_queue_size = 0", "tcp:established = 81950", "tcp:local_reject = 0", "tcp:passive_open = 81948", "tcp:send_timeout = 2", "tcp:sendq_full = 0" ], "id": 12734 } With around 155 registration entries using TCP and TLS in my location database. As you can see, tcp:current_opened_connections = 939 is still pretty high compared to my currently registred users. I have "modparam("registrar", "max_expires", 86400)", because I'm keeping contact entries (even with TCP connection down) for push notifications. I have "tcp_connection_lifetime=3600" configured. Question 1 With "tcp_connection_lifetime=3600", I would expect kamailio to close the established connection after 3600 seconds without traffic. It is pretty obvious that no data has been exchanged over the 4855 established connection during a day. Despite the issue with the Avaya phones is solved automatically after a day, I guess similar stuff or happening, at a different rate, for other users as well. (because current_opened_connections is way higher than registred TCP/TLS users) Question 2 I can list TLS connection with "kamctl rpc tls.list" Can I get a similar list for TCP? (lsof returns a lot of duplicates...) Tks Aymeric -- Antisip - http://www.antisip.com

2 4

Route header
by Duarte Rocha 17 Apr '19

17 Apr '19

Greetings, Is there a difference for Kamailio between these 2 situations. 1 - Method with 2 Route Headers, each one with one address and one uri 2 - Method with 1 Route Header that includes 2 Route URI Best Regards, Duarte Rocha

1 0

allow calls only from registered users
by Gaurav Bmotra 17 Apr '19

17 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i wan to implement user based calling on kamailio , that only can call how registered with user plz help -- *Regards:* Gaurav Kumar

2 1

Send 200 OK response with body to INVITE requests
by David Dean 16 Apr '19

16 Apr '19

Hi all, I'm building an automated load testing system for an rtp media server with TURN that uses Kamailio for SIP. When a client sends an INVITE to Kamailio, I need Kamailio to respond with "200 OK" and the SDP generated by rtpengine. I've enabled auto-answer in the client, so as soon as it receives the response from Kamailio it will start sending RTP packets. Right now I'm using send_reply("200", "OK"); which is sending the right 200 response back to the client, but it doesn't include any body. How can I get Kamailio to send a "200 OK" and add the rtpengine_answer into the body? Appreciate any help, I've been struggling with this for three days. Cheers. rtpengine_offer("SIP-source-address ICE=force-relay RTP");send_reply("200", "OK");

4 7

Getting a new RTP Port on reINVITE during answered call
by Sergio Charrua 16 Apr '19

16 Apr '19

Hi all, I've the following infrastructure (fake IPs) 62.128.128.68 (Telco SBC Inbound) | | 10.19.139.66 (Kamailio + RTPEngine) | | 10.19.139.69 (Asterisk 13) | | 62.128.128.68 (Telco SBC Outbound) Test scenario is: - a mobile calls a destination; the call comes in from Telco SBC Inbound, goes through Kamailio+RTPengine and then to Asterisk which redirects the call to Telco SBC Outbound. - Asterisk box is set with canreinvite=no and nat=force_rport,comedia so it stays in the path when a call comes in SBC Inbound, it is correctly redireted to SBC Outbound. The call is answered, and after 5 or so seconds, SBC Inbound sends a new INVITE to force codecs, which Asterisk replies correctly. the problem is that between Kamailio and SBC Inbund, when Asterisk replies to the second invite, the media port changes, resulting in audio loss: during the call, audio from destination to originator works OK but originator to destination is mute. According to the telco's guys,this is due to the fact that Kamailio is changing during call the media port from original media port to a new one, on reply to the second Invite, and this is wrong and reason why sound from the originator to destination is lost (but not from destination to originator) I'm lost too here..... Any clue? Can anyone help? Resuming I need to make sure that, if during a call some INVITES are sent from SBC to Asterisk, that the reply keeps the same IP_ADDR:port from beginning to the end of the call. As it is today, the second invite changes te RTP Media port and no audi is heard.... If needed I can send PCap files. please find below my kamailio.cfg file #!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* sets the dispatcher list file path */ #!define DS_LIST "/etc/kamailio/dispatcher.list" /* sets the RTP Engine address and port; RTP Engine is a NAT enabled RTP Proxy */ #!define RTP_ENGINE_ADDR "udp:127.0.0.1:60000" // was udp:localhost:60000 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@10.19.139.110/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ //listen=udp:10.19.139.66:5060 listen=udp:eth1:5060 sip_warning=no; ####### Modules Section ######## #set module path mpath="/usr/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.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 "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "path.so" loadmodule "dispatcher.so" loadmodule "rtpengine.so" loadmodule "nathelper.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- 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 we 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 */ 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") //;calltype=$avp(calltype)") # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- modparam("dispatcher", "list_file", DS_LIST) modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("path", "use_received", 1) # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", RTP_ENGINE_ADDR) modparam("nathelper", "received_avp", "$avp(s:rcv)") /************************/ # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) /***********************/ modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:**********@*************/kamailio") ####### Routing Logic ######## # main request routing logic route { xlog("L_INFO","$rm from $si"); xlog("L_DBG","[$fU@$si:$sp]{$rm} {$ru} "); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) // && $fU=="ping" )// && $si=="62.28.173.10" && $fU=="ping")// && strempty((a)ruri.user) && (uri == myself)) { sl_send_reply("200","OK"); exit; } # per request initial checks route(REQINIT); rtpengine_manage(); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()){ route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO","Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ xlog("L_INFO","Recording Route"); record_route(); if (is_method("INVITE")) { if (has_body("application/sdp")) { if (rtpengine_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } } if (is_method("ACK") && has_body("application/sdp")) rtpengine_answer(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } //route(SIPOUT); xlog("L_INFO","Setting PRESENCE"); # handle presence related requests route(PRESENCE); # handle registrations xlog("L_INFO","Handling REGISTRAR"); route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); xlog("L_INFO","**********END*************"); } onreply_route[1] { if (has_body("application/sdp")) rtpengine_answer(); } onreply_route[2] { if (has_body("application/sdp")) rtpengine_offer(); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO","REQINIT Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO","483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } xlog("L_INFO","REQINIT Finishing"); } /***********************/ # Caller NAT detection route[NATDETECT] { xlog("L_INFO","Entering NATDTECT"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO","Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO","Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO","NAT Detect set FLT_NTS = " + FLT_NATS); xlog("L_INFO","Finishing NATDETECT"); return; } /***********************/ # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Entering WITHDLG"); if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO","Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO","Finishing WITHINDLG"); exit; } # Routing to foreign domains route[SIPOUT] { xlog("L_INFO","Entering SIPOUT"); if (uri==myself) { xlog("L_INFO","URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO","Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO","******** RELAY *******"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO","branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_INFO","onreply_route NOT SET!"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_INFO","failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO","t_relay returns FALSE"); sl_reply_error(); } exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO","Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO","Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO","Entering NATMANAGE"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xlog("L_INFO","nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO","NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } xlog("L_INFO","Starting RTPEngine"); rtpengine_manage(); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO","adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { xlog("L_INFO","Entering DISPATCH"); # hash over callerid dispatching on gateways group '1' if(!ds_select_dst("1", "10","4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_INFO","going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } xlog("L_INFO", "Media server $du failed to answer, selecting other one!"); # next DST - only for 500 or local timeout if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) ) { #we mark the destination Inactive and Probing ds_mark_dst("ip"); if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } route[CDRS]{ sql_query("ca","call kamailio_cdrs();","rb"); sql_query("ca","call kamailio_rating('default');","rb"); } Cheers, *Sérgio *

1 0

new RTP Port after INVITE during call
by Sergio Charrua 16 Apr '19

16 Apr '19

Hi all, I've the following infrastructure 62.128.128.68 (Telco SBC Inbound) | | 10.19.139.66 (Kamailio + RTPEngine) | | 10.19.139.69 (Asterisk 13) | | 62.128.128.68 (Telco SBC Outbound) Test scenario is: - a mobile calls a destination; the call comes in from Telco SBC Inbound, goes through Kamailio+RTPengine and then to Asterisk which redirects the call to Telco SBC Outbound. - Asterisk box is set with canreinvite=no and nat=force_rport,comedia so it stays in the path when a call comes in SBC Inbound, it is correctly redireted to SBC Outbound. The call is answered, and after 5 or so seconds, SBC Inbound sends a new INVITE to force codecs, which Asterisk replies correctly. the problem is that between Kamailio and SBC Inbund, when Asterisk replies to the second invite, the media port changes, resulting in audio loss: during the call, audio from destination to originator works OK but originator to destination is mute. According to the telco's guys,this is due to the fact that Kamailio is changing during call the media port from original media port to a new one, on reply to the second Invite, and this is wrong and reason why sound from the originator to destination is lost (but not from destination to originator) I'm lost too here..... Any clue? Can anyone help? Resuming I need to make sure that, if during a call some INVITES are sent from SBC to Asterisk, that the reply keeps the same IP_ADDR:port from beginning to the end of the call. As it is today, the second invite changes te RTP Media port and no audi is heard.... If needed I can send PCap files. please find below my kamailio.cfg file #!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* sets the dispatcher list file path */ #!define DS_LIST "/etc/kamailio/dispatcher.list" /* sets the RTP Engine address and port; RTP Engine is a NAT enabled RTP Proxy */ #!define RTP_ENGINE_ADDR "udp:127.0.0.1:60000" // was udp:localhost:60000 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@10.19.139.110/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ //listen=udp:10.19.139.66:5060 listen=udp:eth1:5060 sip_warning=no; ####### Modules Section ######## #set module path mpath="/usr/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.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 "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "path.so" loadmodule "dispatcher.so" loadmodule "rtpengine.so" loadmodule "nathelper.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- 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 we 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 */ 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") //;calltype=$avp(calltype)") # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- modparam("dispatcher", "list_file", DS_LIST) modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("path", "use_received", 1) # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", RTP_ENGINE_ADDR) modparam("nathelper", "received_avp", "$avp(s:rcv)") /************************/ # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) /***********************/ modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:**********@*************/kamailio") ####### Routing Logic ######## # main request routing logic route { xlog("L_INFO","$rm from $si"); xlog("L_DBG","[$fU@$si:$sp]{$rm} {$ru} "); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) // && $fU=="ping" )// && $si=="62.28.173.10" && $fU=="ping")// && strempty((a)ruri.user) && (uri == myself)) { sl_send_reply("200","OK"); exit; } # per request initial checks route(REQINIT); rtpengine_manage(); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()){ route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO","Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ xlog("L_INFO","Recording Route"); record_route(); if (is_method("INVITE")) { if (has_body("application/sdp")) { if (rtpengine_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } } if (is_method("ACK") && has_body("application/sdp")) rtpengine_answer(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } //route(SIPOUT); xlog("L_INFO","Setting PRESENCE"); # handle presence related requests route(PRESENCE); # handle registrations xlog("L_INFO","Handling REGISTRAR"); route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN //route(PSTN); # dispatch destinations route(DISPATCH); xlog("L_INFO","**********END*************"); } onreply_route[1] { if (has_body("application/sdp")) rtpengine_answer(); } onreply_route[2] { if (has_body("application/sdp")) rtpengine_offer(); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO","REQINIT Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO","483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } xlog("L_INFO","REQINIT Finishing"); } /***********************/ # Caller NAT detection route[NATDETECT] { xlog("L_INFO","Entering NATDTECT"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO","Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO","Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO","NAT Detect set FLT_NTS = " + FLT_NATS); xlog("L_INFO","Finishing NATDETECT"); return; } /***********************/ # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Entering WITHDLG"); if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO","Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO","Finishing WITHINDLG"); exit; } # Routing to foreign domains route[SIPOUT] { xlog("L_INFO","Entering SIPOUT"); if (uri==myself) { xlog("L_INFO","URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO","Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO","******** RELAY *******"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO","branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_INFO","onreply_route NOT SET!"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_INFO","failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO","t_relay returns FALSE"); sl_reply_error(); } exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO","Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO","Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO","Entering NATMANAGE"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xlog("L_INFO","nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO","NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } xlog("L_INFO","Starting RTPEngine"); rtpengine_manage(); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO","adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN xlog("L_INFO","Entering PSTN"); # route to PSTN dialed numbers starting with '+' or '0000' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) { xlog("L_INFO","$rU is not outbound call to PSTN"); return; } if (!starts_with("$si","10.19.139")) { xlog("L_INFO","$rU is not comming from any Media GW"); return; } # only local users allowed to call if(from_uri!=myself && !ds_is_from_list("1") ) { xlog("L_INFO", "from_uri $fu != myself, so this is Not Allowed!!"); sl_send_reply("403", "Not Allowed"); exit; } xlog("L_INFO","$si request"); xlog("L_INFO","$rU is an outbound call to PSTN"); $ru = "sip:" + $rU + "@62.28.173.10"; // + $sel(cfg_get.pstn.gw_ip); route(RELAY); /* if(!ds_select_dst("20", "4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); */ exit; #!endif xlog("L_INFO"," NO PSTN SET!!"); return; } # Dispatch requests route[DISPATCH] { xlog("L_INFO","Entering DISPATCH"); # hash over callerid dispatching on gateways group '1' if(!ds_select_dst("1", "10","4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_INFO","going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Dispatch requests route[DISPATCH_TO_PSTN] { # round robin dispatching on gateways group '20' - PSTN if(!ds_select_dst("20", "10")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } xlog("L_INFO", "Media server $du failed to answer, selecting other one!"); # next DST - only for 500 or local timeout if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) ) { #we mark the destination Inactive and Probing ds_mark_dst("ip"); if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } route[CDRS]{ sql_query("ca","call kamailio_cdrs();","rb"); sql_query("ca","call kamailio_rating('default');","rb"); } Cheers, *Sérgio *

1 0

single Contact per AOR dilema
by Sergiu Pojoga 16 Apr '19

16 Apr '19

Hi ppl, Say we want to allow 1 contact per AOR. modparam("registrar", "max_contacts", 1) Say some UA looses connection before de-registering, then comes back shortly after (before Contact expires) and tries to register with a different Contact, but gets rejected with error "test_max_contacts(): too many contacts for AOR". What's the approach to handle this situation? I'd like the registrar to accept and save that AoR based on the latest registration. Similar to how Asterisk's single-contact stack `chan_sip` works - whoever was the latest to register successfully gets to be the winner. Thanks, --Sergiu

2 2

dlg_onroute(): failed to update dialog lifetime
by Olli Attila 15 Apr '19

15 Apr '19

Hello, I just activated snmpstats module on Kamailio 5.0.7 and added the following line to route(INITIAL) at the bottom of the route like this (For noc group to monitor Kamailio dialogs): # Per SIP request initial checks route[INITIAL] { ... dlg_manage(); } Now I started to gets this error few times in an hour: /usr/sbin/kamailio[22194]: CRITICAL: dialog [dlg_timer.c:200]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7f798ca51b98 tl->next=(nil) tl->prev=(nil) Apr 15 09:54:05 /usr/sbin/kamailio[22194]: ERROR: dialog [dlg_handlers.c:1439]: dlg_onroute(): failed to update dialog lifetime I see no call failures because of this but still interested what is happening here? --Olli Attila

1 0

one way Audio issue
by Gaurav Bmotra 12 Apr '19

12 Apr '19

i have installed kamailio 5.1.x on Ubuntu server 18.4 issue is i am able to connect the call but there is one way audio not both way as i have install rtpproxy also with following config /etc/default/rtpproxy EXTRA_OPTS="-F -s udp:127.0.0.1:7722 -l PUBLIC_IP -d DBUG:LOG_LOCAL0" /etc/kamailio/kamailio.cnf modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") after restart kamailio i got this logs dev /usr/sbin/kamailio[6926]: INFO: rtpproxy [rtpproxy.c:1559]: rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it enabled during calling i got this logs Apr 12 16:05:42 ip-172-31-38-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: rtpproxy [rtpproxy.c:2588]: force_rtp_proxy(): incorre Apr 12 16:05:42 ip-172-31-38-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): dr Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error whi Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: acc [acc.c:477]: acc_db_request(): failed to insert in Apr 12 16:05:43 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13475]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: rtpproxy [rtpproxy.c:2588]: force_rtp_proxy(): incorre Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction ans Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): drive Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: acc [acc.c:477]: acc_db Thanks Gaurav *Regards:* Gaurav Kumar

1 0

Re: [SR-Users] [sr-dev] Audio missing during call
by Mikko Lehto 12 Apr '19

12 Apr '19

Gaurav Bmotra <saigauravmehra91(a)gmail.com> wrote: > HI i have install Kamailio 5.1.2 on Ubuntu 18.4 > there is a audio issue , when we call call from one extension to another ,, > in bigning call goes good , after 9-10 sec it miss audio for one side (one > side can hear other side can't hear ) Hi, mailing list sr-dev is for Kamailio development related topics, please subscribe to sr-users for more general queries. Before posting, make sure you have described in detail your environment and also troubleshooting steps you have done so far: - what modifications you made to default configuration - what happens during SIP transactions in fail scenario - are you using RTP relay or not (rtpproxy/rtpengine) - what have you observed from packet captures Good luck -- Mikko

2 1

command out of sync with db_cluster and db_mysql
by Simone Lazzaris 12 Apr '19

12 Apr '19

Hi everybody; I'm trying to use db_cluster to connect my kamailio installation to a mariadb galera cluster, and it's driving me crazy. In my lab it works without issues, but when I try the setup in the production node, I get some strange errors: Apr 11 14:21:31 ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): driver error on query: Commands out of sync; you can't run this command now (2014) Apr 11 14:21:31 ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query This is my configuration: modparam("db_cluster", "connection","con1=>mysql://openser:PASSWD@X.X.X.30/ openser") modparam("db_cluster", "connection","con2=>mysql://openser:PASSWD@X.X.X.31/ openser") modparam("db_cluster", "connection","con3=>mysql://openser:PASSWD@X.X.X.32/ openser") modparam("db_cluster", "cluster", "cls1=>con1=9r8r;con2=9r8r;con3=9r8r") #!define DBURL "cluster://cls1" #!define SQLOPSURL "database=>cluster://cls1" modparam("dispatcher", "db_url", DBURL) modparam("acc", "db_url", DBURL) modparam("usrloc", "db_url", DBURL) modparam("auth_db", "db_url", DBURL) modparam("permissions", "db_url", DBURL) modparam("alias_db", "db_url", DBURL) modparam( "sqlops", "sqlcon", SQLOPSURL ) I'm using version 5.1.4 of kamailio, freshly compiled from source on a debian 9. Any ideas? *Simone Lazzaris* *Qcom S.p.A.* simone.lazzaris(a)qcom.it[1] | www.qcom.it[2] * LinkedIn[3]* | *Facebook[4]* [5] -------- [1] mailto:simone.lazzaris@qcom.it [2] https://www.qcom.it [3] https://www.linkedin.com/company/qcom-spa [4] http://www.facebook.com/qcomspa [5] https://www.qcom.it/includes/email-banner.gif

2 2

kamailio, asterisk and BLF seems its only using json?
by PICCORO McKAY Lenz 12 Apr '19

12 Apr '19

hi, i try to follow the how to in the wiki https://kb.asipto.com/kamailio:presence:k43-blf and configure my two only phones.. but for any states only one led its blinking, the busy state.. no other state work, (if i connect directly to asterisk all work perfeclty, i using asterisk 13 with kamailio 4.4.4) also read the archived https://lists.kamailio.org/pipermail/sr-users/2013-May/078234.html but no luck wht happened with that how to? it's different for asterisk if are integrated? do i need a pjsip asterisk event the oldes chan? see that in presense module parameters, the server address said that value of Contact header filed for 200 OK replies to SUBSCRIBE and PUBLISH and in NOTIFY messages umm if i have integrated and working asterisk (relatime mode) with kamailio, that address muswt be asterisk? taking in cosideration that i have a route rule said that send auth to asterisk and trust any incomming from it? -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

1 0

having issue updating $td
by Andrew Chen 11 Apr '19

11 Apr '19

Hey guys, Question. I have this block of code at the beginning of route block: if(is_ip($td)) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] IP detected in To domain. Resolving. \n"); exec_avp("host $td | awk '{print substr($$5, 1, length($$5)-1)}'", "$avp(s:newtd)"); xlog("L_INFO", "[$ci][$mi][CSeq $cs] new newtd = $avp(s:newtd)"); $td = $avp(s:newtd); xlog("L_INFO", "[$ci][$mi][CSeq $cs] new td = $td"); } but $td never updates. Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] IP detected in To domain. Resolving. Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] new newtd = sjomaintpsg50.fuzemeeting.com Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] new td = 206.81.191.30 Is there a reason why? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 3

Kamailio v5.1.8 Released
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, Kamailio SIP Server v5.1.8 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.7. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.8 or to v5.2.x series. For more details about version 5.1.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/04/kamailio-v5-1-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is the previous stable branch. The latest stable branch is 5.2, at this time with v5.2.2 being released out of it. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

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Kamailio v5.1.8 Released
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, Kamailio SIP Server v5.1.8 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.7. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.8 or to v5.2.x series. For more details about version 5.1.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/04/kamailio-v5-1-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is the previous stable branch. The latest stable branch is 5.2, at this time with v5.2.2 being released out of it. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

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Releasing Kamailio v5.1.8
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, I am considering to release v5.1.8, an minor version to be built from branch 5.1, on Thursday, this week (April 11, 2019). As usual, check that at least the patches for issues you reported were backported in order to be sure they were not missed during the typical backporting done by developers so far. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 1

Using AWS boto3 with KEMI Python routing scripts
by Michael Loughrey 10 Apr '19

10 Apr '19

Hi All, looking for some advice regarding the proper way to initialise an AWS API boto3 client object for send SNS messages within a KEMI Python 2.7 routing script. I'm sure process forking has a major impact on how this works - from kamailio.cfg fork=yes children=4 - but would greatly appreciate some guidance. I have tried various methods to allocate an client object self.sns_client = boto3.client('sns', region_name=MY_AWS_REGION) 1. within __init__(), only once from module initialisation 2. on demand within a function call by any client code e.g. called from ksr_route_request(), each creating it's own client object and so far all have resulted in intermittent crashes within botocore/client.py ultimately crashing thus : Core was generated by `/usr/local/sbin/kamailio -P /usr/local/kamailio/run/kamailio.pid -f /usr/local/'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f8f9734f754 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 (gdb) bt #0 0x00007f8f9734f754 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #1 0x00007f8f9734f82e in X509_VERIFY_PARAM_free () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #2 0x00007f8f97642f5c in SSL_free () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #3 0x00007f8f86b09c64 in PySSL_dealloc.lto_priv.5 () at ./Modules/_ssl.c:1598 #4 0x00007f8f99cbb007 in insertdict_by_entry (mp=0x7f8f841f1e88, key='_sslobj', hash=<optimized out>, ep=<optimized out>, value=<optimized out>) at ../Objects/dictobject.c:519 #5 0x00007f8f99cbe2cf in insertdict (value=None, hash=1051385741686792393, key='_sslobj', mp=0x7f8f841f1e88) at ../Objects/dictobject.c:556 #6 dict_set_item_by_hash_or_entry (value=None, ep=0x0, hash=1051385741686792393, key='_sslobj', op={'server_hostname': u'sns.us-east-1.amazonaws.com', '_connected': True, '_context': <SSLContext at remote 0x7f8f84232398>, 'server_side': False, '_makefile_refs': 0, '_closed': False, '_sslobj': None, 'do_handshake_on_connect': True, 'suppress_ragged_eofs': True}) at ../Objects/dictobject.c:795 #7 PyDict_SetItem (op=<optimized out>, key=<optimized out>, value=<optimized out>) at ../Objects/dictobject.c:848 #8 0x00007f8f99becec1 in _PyObject_GenericSetAttrWithDict (obj=<optimized out>, name='_sslobj', value=None, dict={'server_hostname': u'sns.us-east-1.amazonaws.com', '_connected': True, '_context': <SSLContext at remote 0x7f8f84232398>, 'server_side': False, '_makefile_refs': 0, '_closed': False, '_sslobj': None, 'do_handshake_on_connect': True, 'suppress_ragged_eofs': True}) at ../Objects/object.c:1529 #9 0x00007f8f99bed437 in PyObject_SetAttr ( v=<SSLSocket(server_hostname=u'sns.us-east-1.amazonaws.com', _connected=True, _context=<SSLContext at remote 0x7f8f84232398>, server_side=False, _makefile_refs=0, _closed=False, _sslobj=None, do_handshake_on_connect=True, suppress_ragged_eofs=True) at remote 0x7f8f841f62a8>, name=<optimized out>, value=None) at ../Objects/object.c:1247 Here's my setup host : Debian9/stretch python --version Python 2.7.13 kamailio -v version: kamailio 5.2.2 (x86_64/linux) 67f967-dirty flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, MEM_JOIN_FREE ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 67f967 -dirty compiled on 23:44:31 Apr 8 2019 with gcc 6.3.0 Cheers Mike

2 3

Rtpengine - no RTP packets going out
by Istvan Mogyorosi 10 Apr '19

10 Apr '19

Dear all, This is my first post after reading a lot in this mailing-list. I'm trying to use Kamailio 5.1 with the dispatcher module and rtpengine acting as SIP + RTP proxy. I have 6 asterisk servers in a private subnet that should talk with the peer via a single IP like this: Asterisk 1..n|---> | GW.PRIVATE.IP -o- GW.PUBLIC.IP |----> PEER.SIP.TRUNK I'm on Centos 7, with firewalld configured, iptables module is loaded and the rule is well defined. Packet forwarding is also enabled. Chain rtpengine (1 references) target prot opt source destination RTPENGINE udp -- anywhere anywhere RTPENGINE id:40 My call flow seems to be fine, Kamailio/rtpengine private IP is the outboundproxy parameter of Asterisk instances. My problem is that RTP packets are not present on the public interface, the rtpengine final log showing the 2 sessions, but I'm not sure this is what I want or simply the firewall does not let it out ? (To be more precise PEER.SIP.TRUNK is the trunk for SIP traffic, I have multiple IP addresses for media to connect to, reinvites are allowed) Closing call due to timeout Final packet stats: --- Tag 'as6d12caea', created 1:00 ago for branch '', in dialogue with 'as541b1e61' ------ Media #1 (audio over RTP/AVP) using unknown codec --------- Port GW.PRIVATE.IP:10000 <> 192.168.30.13:11152, SSRC 0, 0 p, 0 b, 0 e, 60 ts --------- Port GW.PRIVATE.IP:10001 <> 192.168.30.13:11153 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 60 ts --- Tag 'as541b1e61', created 1:00 ago for branch '', in dialogue with 'as6d12caea' ------ Media #1 (audio over RTP/AVP) using unknown codec --------- Port GW.PUBLIC.IP:10000 <> PEER.SIP.TRUNK:28216, SSRC 0, 0 p, 0 b, 0 e, 60 ts --------- Port GW.PUBLIC.IP:10001 <> PEER.SIP.TRUNK:28217 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 60 ts Best regards, Istvan

2 2

Kamailio and CDR
by Yu Boot 10 Apr '19

10 Apr '19

Is there any other way to retrieve CDRs from Kami other than dialog/acc modules and writing CDRs to syslog only? Thanks.

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Help request in regards to Kamailio
by Anas ZAHHAF 10 Apr '19

10 Apr '19

Dear, I am writing to ask for help, I found your Email after I've subscribed to Kamailio.org. I am a beginner in Linux environment ( I worked on Windows for 20 years), it would be very helpful if you can help me installing Kamailio on a computer running Debian and how to use the load balancing module so that I can split the charge between 2 asterisk servers, by sharing any useful tutorial or document or even by Teamviewer if this doesn't disturb you. Please help me, all the tutorials I find on internet aren't made for beginners. I will be awaiting your reply and excuse me again if I disturb you. Best regards, -- *Anas ZAHHAF* *105, rue des Moines75017 PARIS* *+33(0)7.66.31.31.85* azahhaf(a)zaion.ai <niluse(a)zaion.ai> | www.zaion.ai ᐧ

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change failure status if reason is q.850 cause=31
by Techinnovation 10 Apr '19

10 Apr '19

Hello, How can I change the status where Reason: q.850;cause=31. ? This code change status for all where we receive status 480 to 404. Is it possible to this code also to add second condition - if reason is like "q.850;cause=31" failure_route[MANAGE_FAILURE] { ... if (t_check_status("480")) { t_reply("404","Unallocated number"); exit; } Need to modify this massage. SIP/2.0 480 Temporarily Not Available. Allow: UPDATE,REFER,INFO. Call-ID: 6b11b153405a5103330697b774b97f5a@192.168.2.2:5060. Contact: <sip:192.168.3.3:5060>. CSeq: 102 INVITE. From: "380668375820" <sip:380668375820@192.168.2.2>;tag=as445242c0. Reason: q.850;cause=31. Server: Cirpack/v4.76 (gw_sip). To: <sip:9052946032@192.168.3.3:5060>;tag=00-07929-6e4a3079-75447e2a5. Content-Length: 0. Thanks. BR, Aleksandr

2 1

Question regarding retcode
by Andrew Chen 09 Apr '19

09 Apr '19

Hey all, Question regarding the type of retcode. I ran a curl command and $retcode returned 200. I check against that variable if it's 200 or not. I tried to check 200 as a string and integer but never matches so I ran these lines of code: xlog("L_INFO", "[$ci][$mi][CSeq $cs] Validating meeting ($rU). Result: $var(result) Return code: $retcode \n"); if(is_int($retcode)) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is an integer \n"); } if(typeof("$retcode", "str")) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is a string \n"); } None of these xlog outputed. How do I check for $retcode in if or switch statement? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Centos 7 opensuse repo issues for 5.2
by Jeffrey Gray 09 Apr '19

09 Apr '19

Good afternoon, We are unable to get kamailio 5.2 to install using the recommended opensuse repos for Centos 7. Is there any way to resolve the following errors: Thanks very much for your assistance! using the following repo setup in /etc/yum.repos.d/ [home_kamailio_v5.2.x-rpms] name=RPM Packages for Kamailio v5.2.x (CentOS_7) type=rpm-md baseurl=http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rp… gpgcheck=1 gpgkey=http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpm… enabled=1 Resolving Dependencies --> Running transaction check ---> Package kamailio.x86_64 0:5.2.2-3.el7.centos will be installed --> Finished Dependency Resolution Dependencies Resolved ============================================================================================================================================================================= Package Arch Version Repository Size ============================================================================================================================================================================= Installing: kamailio x86_64 5.2.2-3.el7.centos home_kamailio_v5.2.x-rpms 5.6 M Transaction Summary ============================================================================================================================================================================= Install 1 Package Total download size: 5.6 M Installed size: 25 M Downloading packages: kamailio-5.2.2-3.el7.centos.x8 FAILED ================- ] 2.3 MB/s | 1.7 MB 00:00:01 ETA http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.2.x-rpms clean metadata Trying other mirror. Error downloading packages: kamailio-5.2.2-3.el7.centos.x86_64: [Errno 256] No more mirrors to try. If we run the suggested clean metadata command, then the error changes to: sudo yum -y install kamailio Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirrors.advancedhosters.com * epel: mirror.cogentco.com * extras: mirrors.advancedhosters.com * updates: mirrors.advancedhosters.com Resolving Dependencies --> Running transaction check ---> Package kamailio.x86_64 0:5.2.2-3.el7.centos will be installed --> Finished Dependency Resolution Dependencies Resolved ============================================================================================================================================================================= Package Arch Version Repository Size ============================================================================================================================================================================= Installing: kamailio x86_64 5.2.2-3.el7.centos home_kamailio_v5.2.x-rpms 5.6 M Transaction Summary ============================================================================================================================================================================= Install 1 Package Total download size: 5.6 M Installed size: 25 M Downloading packages: Delta RPMs disabled because /usr/bin/applydeltarpm not installed. kamailio-5.2.2-3.el7.centos.x8 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not Satisfiable Trying other mirror. Error downloading packages: kamailio-5.2.2-3.el7.centos.x86_64: [Errno 256] No more mirrors to try.

3 3

Upstream ignores reply, because To-tag doesn't match?
by Sergiu Pojoga 09 Apr '19

09 Apr '19

Hi ppl, Scenario: invite from upstream is t_relayed to a client gateway. After `fr_inv_timeout` occurs, in a failure route I simply t_reply with "503 - Service unavailable", then exit(). However, the upstream provider simply ignores this reply, call doesn't hang up. This doesn't happen when the client's gateway generates identical 503 or other negative replies. I suspect this is happening because the To-tag in the Kam generated 503 reply doesn't match with the To-tag that previously was forwarded from the client's gateway to the upstream provider in the `180 Ringing`. Any suggestions what can be done about it? 2019/04/06 10:42:46.127177 65.39.xxx.xxx:5060 -> 208.72.xxx.xxx:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 208.72.xxx.xxx:5060;received=208.72.xxx.xxx;rport=5060;branch=z9hG4bK1982422573 Record-Route: <sip:65.39.xxx.xxx;lr=on;did=915.8af1> From: "SERGIU" <sip:514XXXXXXX@208.72.xxx.xxx>;isup-oli=00;tag=1975755942 To: <sip:514XXXXXXX@65.39.xxx.xxx>;tag=as2ab54180 Call-ID: DID-28270826(a)208.72.xxx.xx CSeq: 477023 INVITE Supported: replaces, timer, path Contact: <sip:1514XXXXXXX@65.39.xxx.xxx:5060> Content-Length: 0 2019/04/06 10:42:51.072516 65.39.xxx.xxx:5060 -> 208.72.xxx.xxx:5060 SIP/2.0 503 Service Unavailable Call-ID: DID-28270826(a)208.72.xxx.xx Via: SIP/2.0/UDP 208.72.xxx.xxx:5060;rport=5060;branch=z9hG4bK1982422573;received=208.72.xxx.xxx From: "SERGIU" <sip:514XXXXXXX@208.72.xxx.xxx>;isup-oli=00;tag=1975755942 To: <sip:514XXXXXXX@65.39.xxx.xxx>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8e82 CSeq: 477023 INVITE Server: KAM Content-Length: 0 Thanks, --Sergiu

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