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]> 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 -> 10.1.1.4:5060 > INVITE sip:[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]>;tag=as5ce97f3d > To: <sip:[email protected];user=phone> > Contact: <sip:[email protected]:5060> > Call-ID: [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 -> 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]>;tag=as5ce97f3d > To: <sip:[email protected];user=phone> > Call-ID: [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 -> 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]>;tag=as5ce97f3d > To: <sip:[email protected] > ;user=phone>;tag=e68db714ad3ba80833ca2c670d982872.aebb > Call-ID: [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]> 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]> 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) $ci} " >>> children=1 >>> >>> server_id = 10 >>> xavp_via_params = "via" >>> disable_tcp=yes >>> auto_aliases=no >>> listen=udp:0.0.0.0:5060 >>> >>> ####### Modules Section ######## >>> >>> loadmodule "jsonrpcs.so" >>> loadmodule "kex.so" >>> loadmodule "corex.so" >>> loadmodule "tm.so" >>> loadmodule "tmx.so" >>> loadmodule "sl.so" >>> loadmodule "rr.so" >>> loadmodule "pv.so" >>> loadmodule "maxfwd.so" >>> loadmodule "textops.so" >>> loadmodule "siputils.so" >>> loadmodule "xlog.so" >>> loadmodule "sanity.so" >>> loadmodule "ctl.so" >>> loadmodule "cfg_rpc.so" >>> loadmodule "acc.so" >>> loadmodule "counters.so" >>> >>> # ----------------- setting module-specific parameters --------------- >>> >>> >>> # ----- jsonrpcs params ----- >>> modparam("jsonrpcs", "pretty_format", 1) >>> modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") >>> modparam("jsonrpcs", "dgram_socket", >>> "/var/run/kamailio/kamailio_rpc.sock") >>> modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") >>> >>> # ----- tm params ----- >>> modparam("tm", "failure_reply_mode", 3) >>> modparam("tm", "fr_timer", 30000) >>> modparam("tm", "fr_inv_timer", 120000) >>> modparam("rr", "enable_full_lr", 0) >>> modparam("rr", "append_fromtag", 0) >>> modparam("acc", "early_media", 0) >>> modparam("acc", "report_ack", 0) >>> modparam("acc", "report_cancels", 0) >>> modparam("acc", "detect_direction", 0) >>> modparam("acc", "log_flag", FLT_ACC) >>> modparam("acc", "log_missed_flag", FLT_ACCMISSED) >>> modparam("acc", "log_extra", >>> "src_user=$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] >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> [email protected] >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > > _______________________________________________ > Kamailio (SER) - Users Mailing > [email protected]https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- > Henning Westerholt - https://skalatan.de/blog/ > Kamailio services - https://skalatan.de/services > >
_______________________________________________ Kamailio (SER) - Users Mailing List [email protected] https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
