Re: [SR-Users] Reverse Proxy Asterisk - SIP Provider

I did test without the drop(); same result. Will try with this https://kamailio.org/docs/modules/5.2.x/modules/dispatcher.html#dispatcher.…, 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 &lt;hw(a)skalatan.de&gt; 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.… > > 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 &lt;hw(a)skalatan.de&gt; > 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:+40XXXXXXXXX@10.1.1.4;user=phone SIP/2.0 >> Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK3ecba174;rport >> Max-Forwards: 70 >> From: "test" <sip:+40YYYYYYYY@10.1.1.10>;tag=as5ce97f3d >> To: <sip:+40XXXXXXXXX@10.1.1.4;user=phone> >> Contact: <sip:+40YYYYYYYY@10.1.1.10:5060> >> Call-ID: 7c1169bf0b7a09472dd455d76b6c12bb(a)10.1.1.10 >> 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:+40YYYYYYYY@10.1.1.10>;tag=as5ce97f3d >> To: <sip:+40XXXXXXXXX@10.1.1.4;user=phone> >> Call-ID: 7c1169bf0b7a09472dd455d76b6c12bb(a)10.1.1.10 >> 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:+40YYYYYYYY@10.1.1.10>;tag=as5ce97f3d >> To: <sip:+40XXXXXXXXX@10.1.1.4 >> ;user=phone>;tag=e68db714ad3ba80833ca2c670d982872.aebb >> Call-ID: 7c1169bf0b7a09472dd455d76b6c12bb(a)10.1.1.10 >> 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 &lt;pojogas(a)gmail.com&gt; 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 &lt;cezar(a)mokalife.ro&gt; 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 >>>> sr-users(a)lists.kamailio.org >>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://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 > >