[SR-Users] Reverse Proxy Asterisk - SIP Provider

[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 at 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 at 10.1.1.10>;tag=as5ce97f3d
To: <sip:+40XXXXXXXXX at 10.1.1.4;user=phone>
Contact: <sip:+40YYYYYYYY at 10.1.1.10:5060>
Call-ID: 7c1169bf0b7a09472dd455d76b6c12bb at 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 at 10.1.1.10>;tag=as5ce97f3d
To: <sip:+40XXXXXXXXX at 10.1.1.4;user=phone>
Call-ID: 7c1169bf0b7a09472dd455d76b6c12bb at 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 at 10.1.1.10>;tag=as5ce97f3d
To: <sip:+40XXXXXXXXX at 10.1.1.4
;user=phone>;tag=e68db714ad3ba80833ca2c670d982872.aebb
Call-ID: 7c1169bf0b7a09472dd455d76b6c12bb at 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 at 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 at 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) $ci} "
>> children=1
>>
>> server_id = 10
>> xavp_via_params = "via"
>> disable_tcp=yes
>> auto_aliases=no
>> listen=udp:0.0.0.0:5060
>>
>> ####### Modules Section ########
>>
>> loadmodule "jsonrpcs.so"
>> loadmodule "kex.so"
>> loadmodule "corex.so"
>> loadmodule "tm.so"
>> loadmodule "tmx.so"
>> loadmodule "sl.so"
>> loadmodule "rr.so"
>> loadmodule "pv.so"
>> loadmodule "maxfwd.so"
>> loadmodule "textops.so"
>> loadmodule "siputils.so"
>> loadmodule "xlog.so"
>> loadmodule "sanity.so"
>> loadmodule "ctl.so"
>> loadmodule "cfg_rpc.so"
>> loadmodule "acc.so"
>> loadmodule "counters.so"
>>
>> # ----------------- setting module-specific parameters ---------------
>>
>>
>> # ----- jsonrpcs params -----
>> modparam("jsonrpcs", "pretty_format", 1)
>> modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo")
>> modparam("jsonrpcs", "dgram_socket",
>> "/var/run/kamailio/kamailio_rpc.sock")
>> modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl")
>>
>> # ----- tm params -----
>> modparam("tm", "failure_reply_mode", 3)
>> modparam("tm", "fr_timer", 30000)
>> modparam("tm", "fr_inv_timer", 120000)
>> modparam("rr", "enable_full_lr", 0)
>> modparam("rr", "append_fromtag", 0)
>> modparam("acc", "early_media", 0)
>> modparam("acc", "report_ack", 0)
>> modparam("acc", "report_cancels", 0)
>> modparam("acc", "detect_direction", 0)
>> modparam("acc", "log_flag", FLT_ACC)
>> modparam("acc", "log_missed_flag", FLT_ACCMISSED)
>> modparam("acc", "log_extra",
>> "src_user=$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 at lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20190725/b459c346/attachment.html>