sr-users April 2019

sr-users@lists.kamailio.org

81 participants
109 discussions

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

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sr-users April 2019