If you didn’t get the answer you wanted on the users list, posting to the dev list is not the socially correct alternative strategy. The dev list is for project development issues. Moreover, you are reposting an identical question, suggesting that you did not consider the answers you have already received on the users list them.

Sent from mobile, with due apologies for brevity and errors.

On Mar 30, 2020, at 3:27 AM, egemen ulus <ulus_egemen@hotmail.com> wrote:


Hi All,

I'm trying to establish a topology with Kamailio which is located between Asterisk and Outside.

My issue is "transaction matching failed" (t_check_trans failed). That means Kamailio does not recognise ACK & BYE message to forwards. I had a check on email list and google but found nothing.
I'm using it with record_route.

SIP signalling and part of kamailio.cfg is below. Is there any suggestion?



route[WITHINDLG] {
if (!has_totag()) return;
if (loose_route()) {
route(DLGURI);
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
}
else if ( is_method("ACK") ) {
xlog("ack is came: $si:$sp\n");
route(NATMANAGE);
}
else if ( is_method("NOTIFY") ) {
record_route();
}
route(RELAY);
exit;
}
if (is_method("SUBSCRIBE") && uri == myself) {
route(PRESENCE);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
route(RELAY);
exit;
} else {
exit;
}
}
sl_send_reply("404","Not here");
exit;
}

route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}

route(RTPPROXY);

if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
if(is_first_hop())
set_contact_alias();
}
}
#!endif
return;
}





Asterisk > Kamailio

INVITE sip:user@outside_ip SIP/2.0
Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK1b56cce0
Max-Forwards: 70
From: "KUYRUK" <sip:0553847aaaa@asterisk_ip>;tag=as3dde7cdb
To: <sip:user@outside_ip>
Contact: <sip:0553847aaaa@asterisk_ip:5060>
Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.32.3


Kamailio > Outside

INVITE sip:user@outside_ip SIP/2.0
Record-Route: <sip:kamailio_ip;lr;nat=yes>
Via: SIP/2.0/UDP kamailio_ip;branch=z9hG4bK7b3f.8e4716eeea4b0dc45cc51d21b5aded57.0
Via: SIP/2.0/UDP asterisk_ip:5060;rport=5060;branch=z9hG4bK1b56cce0
From: "KUYRUK" <sip:0553847aaaa@asterisk_ip>;tag=as3dde7cdb
To: <sip:user@outside_ip>
Contact: <sip:0553847aaaa@asterisk_ip:5060;alias=asterisk_ip~5060~1>
Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.32.3


Outside > Kamailio

SIP/2.0 200 OK
Via: SIP/2.0/UDP kamailio_ip;branch=z9hG4bK7b3f.8e4716eeea4b0dc45cc51d21b5aded57.0
Via: SIP/2.0/UDP asterisk_ip:5060;rport=5060;branch=z9hG4bK1b56cce0
From: "KUYRUK" <sip:0553847aaaa@asterisk_ip>;tag=as3dde7cdb
To: <sip:user@outside_ip>;tag=kv8iml6ur9
Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060
CSeq: 102 INVITE
Record-Route: <sip:kamailio_ip;lr;nat=yes>
Contact: <sip:user@outside_ip:5060;alias=x.x.x.x~18005~6;transport=udp>


Kamailio > Asterisk

SIP/2.0 200 OK
Via: SIP/2.0/UDP asterisk_ip:5060;rport=5060;branch=z9hG4bK1b56cce0
From: "KUYRUK" <sip:0553847aaaa@asterisk_ip>;tag=as3dde7cdb
To: <sip:user@outside_ip>;tag=kv8iml6ur9
Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060
CSeq: 102 INVITE
Record-Route: <sip:kamailio_ip;lr;nat=yes>
Contact: <sip:user@outside_ip:5060;alias=x.x.x.x~18005~6;transport=udp>


Asterisk > Kamailio THIS ACK IS NOT ROUTED TO OUTSIDE BY KAMAILIO!!

ACK sip:user@outside_ip:5060;alias=x.x.x.x~18005~6;transport=udp SIP/2.0
Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK73366924
Route: <sip:kamailio_ip;lr;nat=yes>
From: "KUYRUK" <sip:0553847aaaa@asterisk_ip>;tag=as3dde7cdb
To: <sip:user@outside_ip>;tag=kv8iml6ur9
Contact: <sip:0553847aaaa@asterisk_ip:5060>
Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.32.3
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