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
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