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

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.e...

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 <hw@skalatan.demailto:hw@skalatan.de> 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 10.1.1.10:5060http://10.1.1.10:5060 -> 10.1.1.4:5060http://10.1.1.4:5060 INVITE sip:+40XXXXXXXXX@10.1.1.4mailto:sip%3A%2B40XXXXXXXXX@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@10.1.1.10mailto:sip%3A%2B40YYYYYYYY@10.1.1.10>;tag=as5ce97f3d To: <sip:+40XXXXXXXXX@10.1.1.4mailto:sip%3A%2B40XXXXXXXXX@10.1.1.4;user=phone> Contact: <sip:+40YYYYYYYY@10.1.1.10:5060http://sip:+40YYYYYYYY@10.1.1.10:5060> Call-ID: 7c1169bf0b7a09472dd455d76b6c12bb@10.1.1.10mailto:7c1169bf0b7a09472dd455d76b6c12bb@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 10.1.1.4:5060http://10.1.1.4:5060 -> 10.1.1.10:5060http://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@10.1.1.10mailto:sip%3A%2B40YYYYYYYY@10.1.1.10>;tag=as5ce97f3d To: <sip:+40XXXXXXXXX@10.1.1.4mailto:sip%3A%2B40XXXXXXXXX@10.1.1.4;user=phone> Call-ID: 7c1169bf0b7a09472dd455d76b6c12bb@10.1.1.10mailto:7c1169bf0b7a09472dd455d76b6c12bb@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 10.1.1.4:5060http://10.1.1.4:5060 -> 10.1.1.10:5060http://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@10.1.1.10mailto:sip%3A%2B40YYYYYYYY@10.1.1.10>;tag=as5ce97f3d To: <sip:+40XXXXXXXXX@10.1.1.4mailto:sip%3A%2B40XXXXXXXXX@10.1.1.4;user=phone>;tag=e68db714ad3ba80833ca2c670d982872.aebb Call-ID: 7c1169bf0b7a09472dd455d76b6c12bb@10.1.1.10mailto:7c1169bf0b7a09472dd455d76b6c12bb@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 <pojogas@gmail.commailto:pojogas@gmail.com> 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 <cezar@mokalife.romailto:cezar@mokalife.ro> 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:0.0.0.0:5060http://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@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org 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