Hello Mihai, the screenshot tells that the INVITE is routed to localhost:5060. Is your asterisk actually listening on localhost? About the previous question about rewriting headers - you should get at least a response if your request is ending up somehow in the asterisk, and/or see something in the asterisk logs.
Cheers, Henning Am 02.08.19 um 11:54 schrieb Mihai Cezar: Here is another SIP trace from Kamailio to Asterisk (localhost) - both are on the same machine, and still get it to respond to Invite (getting timeouts). In asterisk logs i did not find anything. the output sysctl is this: net.ipv4.ip_forward = 1 net.ipv4.conf.all.send_redirects = 0 net.ipv4.conf.default.send_redirects = 0 net.ipv4.conf.all.secure_redirects = 0 net.ipv4.conf.default.secure_redirects = 0 net.ipv4.conf.all.accept_redirects = 0 net.ipv4.conf.default.accept_redirects = 0 net.ipv4.conf.all.accept_source_route = 0 net.ipv4.conf.default.accept_source_route = 0 net.ipv4.icmp_ignore_bogus_error_responses = 1 net.ipv4.icmp_echo_ignore_broadcasts = 1 net.ipv4.conf.all.log_martians = 0 net.ipv4.conf.default.rp_filter = 0 net.ipv4.conf.all.rp_filter = 0 net.ipv4.tcp_syncookies = 1 kernel.sysrq = 0 kernel.core_uses_pid = 1 net.ipv4.tcp_syncookies = 1 kernel.msgmnb = 65536 kernel.msgmax = 65536 kernel.shmmax = 68719476736 kernel.shmall = 4294967296 [...] On Tue, Jul 30, 2019 at 5:30 PM Mihai Cezar <[email protected]<mailto:[email protected]>> wrote: I tried with the example bellow, and it's forwarding the invite, but, i get request timeout, maybe i need to rewrite some contact headers? Trace for Incoming via sip_provider -> proxy -> asterisk: This is my version: kamailio 5.2.3 (x86_64/linux) c36229 2019/07/30 17:23:18.258880 KAMAILIO_IP:5060 -> ASTERISK_IP:5060 INVITE sip:+XXXXXXXXXXX@KAMAILIO_IP:5060;user=phone SIP/2.0 Record-Route: <sip:KAMAILIO_IP;lr=on> Via: SIP/2.0/UDP KAMAILIO_IP;branch=z9hG4bK5bf9.bf37af911cb7c8db93893a9ac5e160d3.0 Via: SIP/2.0/UDP PROVIDER_IP:5060;branch=z9hG4bKhfc5o810bggumum4q6g0.1 From: <sip:+ZZZZZZZZZZZ@PROVIDER_IP;user=phone>;tag=01011555152057 To: <sip:[email protected]<mailto:sip%3a%[email protected]>;user=phone> Max-Forwards: 68 Call-ID: jibr21823173009wbdghefelkf...@clu05.mscs.mnc010.mcc226.3gppnetwork.org<mailto:jibr21823173009wbdghefelkf...@clu05.mscs.mnc010.mcc226.3gppnetwork.org> CSeq: 58311169 INVITE P-Asserted-Identity: <sip:+ZZZZZZZZZZZ@PROVIDER_IP;user=phone> Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE P-Charging-Vector: icid-value=0D3D702000-0730-17231801;icid-generated-at=CLU05.MSCS.MNC010.MCC226.3GPPNETWORK.ORG<http://CLU05.MSCS.MNC010.MCC226.3GPPNETWORK.ORG>;orig-ioi=MSCS.MNC010.MCC226.3GPPNETWORK.ORG<http://MSCS.MNC010.MCC226.3GPPNETWORK.ORG> P-Early-Media: supported Supported: 100rel,histinfo Content-Type: application/sdp Contact: <sip:+ZZZZZZZZZZZ@PROVIDER_IP:5060;transport=udp> Content-Length: 561 v=0 o=- 14284082 14284082 IN IP4 PROVIDER_IP s=- c=IN IP4 PROVIDER_IP t=0 0 a=sendrecv m=audio 56884 RTP/AVP 96 97 98 8 18 99 c=IN IP4 PROVIDER_IP b=RR:0 b=RS:0 a=rtpmap:96 AMR/8000 a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0;octet-align=1 a=rtpmap:97 AMR/8000 a=fmtp:97 mode-set=7;max-red=0;octet-align=1 a=rtpmap:98 GSM-EFR/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-15 a=ptime:20 a=maxptime:40 ------------------------------------------------------------------------------------------------ #!KAMAILIO # # *** To run in debug mode: # - define WITH_DEBUG # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 disable_tcp=yes auto_aliases=no port=5060 sip_warning=no ####### Modules Section ######## #loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- 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", "log_flag", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 0) ####### Routing Logic ######## # main request routing logic request_route { # per request initial checks route(REQINIT); # 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(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if (is_method("OPTIONS")) { sl_send_reply("200","Keepalive"); exit; } # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; sl_send_reply("404", "No registrar"); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No SIP destionations alive"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } On Fri, Jul 26, 2019 at 12:28 AM Mihai Cezar <[email protected]<mailto:[email protected]>> wrote: I did test without the drop(); same result. Will try with this https://kamailio.org/docs/modules/5.2.x/modules/dispatcher.html#dispatcher.ex.config, but i don't use register the sip trunk using IP authentication (there is no nat in this scenario, all private IPs I wanted to avoid rtpproxy/rtpengine). Thanks! On Thu, Jul 25, 2019 at 11:51 PM Henning Westerholt <[email protected]<mailto:[email protected]>> wrote: Hi Mihai, indeed sounds like this. :-) There is a drop call in your cfg in the path that is taken in your sip trace. This will cause a drop of the message that is currently processed. The default cfg is a bit larger with all the #!ifdef cases, and maybe a bit difficult to understand. Have you tried this cfg: https://kamailio.org/docs/modules/5.2.x/modules/dispatcher.html#dispatcher.ex.config This is just a simple kamailio dispatcher cfg (stateful forwarding and record routing). Just add your asterisk server to a dispatcher.list file and it should work. This cfg will block REGISTER and presence requests, but you can easily deactivate it. Cheers, Henning Am 25.07.19 um 21:12 schrieb Mihai Cezar: No, it dosen't forward it to the SIP provider, it basicaly loops, i am guessing that my config its the problem... On Thu, Jul 25, 2019 at 9:53 PM Henning Westerholt <[email protected]<mailto:[email protected]>> wrote: Hello Mihai, your trace just shows the INVITE, 100, 183. There is no 200 OK, therefore also no ACK. Is there some thing missing? Does the called side actually accept the call? Cheers, Henning Am 25.07.19 um 19:23 schrieb Mihai Cezar: Well, i've tried both opensips and kamailio but with kamailio i got the most far. Bellow it's a trace of an outgoing call, the trace is from kamailio box. Legend: 10.1.1.10 is the Asterisk Box, 10.1.1.4 is Kamailio. 2019/07/25 20:16:24.479179 10.1.1.10:5060<http://10.1.1.10:5060> -> 10.1.1.4:5060<http://10.1.1.4:5060> INVITE sip:[email protected]<mailto:sip%3A%[email protected]>;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK3ecba174;rport Max-Forwards: 70 From: "test" <sip:[email protected]<mailto:sip%3A%[email protected]>>;tag=as5ce97f3d To: <sip:[email protected]<mailto:sip%3A%[email protected]>;user=phone> Contact: <sip:[email protected]:5060<http://sip:[email protected]:5060>> Call-ID: [email protected]<mailto:[email protected]> CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 25 Jul 2019 17:16:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 238 2019/07/25 20:16:24.482259 10.1.1.4:5060<http://10.1.1.4:5060> -> 10.1.1.10:5060<http://10.1.1.10:5060> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK3ecba174;rport=5060;received=10.1.1.10 From: "test" <sip:[email protected]<mailto:sip%3A%[email protected]>>;tag=as5ce97f3d To: <sip:[email protected]<mailto:sip%3A%[email protected]>;user=phone> Call-ID: [email protected]<mailto:[email protected]> CSeq: 102 INVITE Server: kamailio (5.2.3 (x86_64/linux)) Content-Length: 0 2019/07/25 20:16:24.482385 10.1.1.4:5060<http://10.1.1.4:5060> -> 10.1.1.10:5060<http://10.1.1.10:5060> SIP/2.0 183 Outgoing session to Avoxi Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK3ecba174;rport=5060;received=10.1.1.10 From: "test" <sip:[email protected]<mailto:sip%3A%[email protected]>>;tag=as5ce97f3d To: <sip:[email protected]<mailto:sip%3A%[email protected]>;user=phone>;tag=e68db714ad3ba80833ca2c670d982872.aebb Call-ID: [email protected]<mailto:[email protected]> CSeq: 102 INVITE Server: kamailio (5.2.3 (x86_64/linux)) Content-Length: 0 On Thu, Jul 25, 2019 at 7:53 PM Sergiu Pojoga <[email protected]<mailto:[email protected]>> wrote: Have you tried changing the trunk's name from opensips-trunk to kamailio-trunk? On the serious side, a SIP trace would help. On Thu, Jul 25, 2019 at 12:26 PM Mihai Cezar <[email protected]<mailto:[email protected]>> wrote: Hi all, I've tried to create a reverse proxy to forward incoming request that came from SIP provider to Asterisk PBX and forward the requests from asterisk to kamailio then sip provider. What i get is that I see the invite, but is like no ACK. Thanks in advance. M kamailio.cfg: #!KAMAILIO # ####### Defined Values ######### # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR debug=3 log_stderror=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " children=1 server_id = 10 xavp_via_params = "via" disable_tcp=yes auto_aliases=no listen=udp:0.0.0.0:5060<http://0.0.0.0:5060> ####### Modules Section ######## loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 0) modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) ####### Routing Logic ######## request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE|REFER")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); sl_send_reply("100","Trying"); if ($si == "172.16.16.1") { sl_send_reply("183","Incoming session from Avoxi"); rewritehost("10.1.1.10"); #exit; } else if ($si == "10.1.1.10"){ # receiving response from client sl_send_reply("183","Outgoing session to Avoxi"); #rewritehost("172.16.16.1"); drop; exit; } else { sl_send_reply("500","No configured IP!"); drop; exit; } } if ($rU==$null) { sl_send_reply("484","Address Incomplete"); exit; } # received from main server - send to client and add via tokens for anycast handling via_add_srvid("1"); $xavp(via=>node) = "10.1.1.4"; via_add_xavp_params("1"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } if ($si == "10.1.1.4") { xlog("L_WARN", "$ci|end|dropping message"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); } else if ( is_method("NOTIFY") ) { record_route(); } route(RELAY); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { route(RELAY); exit; } else { exit; } } sl_send_reply("400","Loop detected"); exit; } # TM manage for outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); } # TM manage for incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); } # TM manage for failure routing cases failure_route[MANAGE_FAILURE] { if (t_is_canceled()) exit; } asterisk - sip.conf [opensips-trunk](sip-provider) fromdomain=10.1.1.10 host=10.1.1.4 context=from-trunk type=friend insecure=invite,port trunk=yes _______________________________________________ Kamailio (SER) - Users Mailing List [email protected]<mailto:[email protected]> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List [email protected]<mailto:[email protected]> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List [email protected]<mailto:[email protected]> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services _______________________________________________ Kamailio (SER) - Users Mailing List [email protected]<mailto:[email protected]> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services
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