I am not sure about your situation, but in my case - Asterisk respond to any message, and wrong paths were corrected the way I showed.
By the way - I am using also rtpproxy. Although it should not interfere here at all.
I used wireshark on Asterisk and on Kamailio servers to find what exactly happens.
The idea is - I check  "target" (for ACK and BYE) and if target is Kamailio server, I forward package to Asterisk.
As I mentioned - I am not sure what exactly is wrong - with my setup, or Kamailio or Asterisk - but my go around works well for me.


On Fri, Oct 25, 2013 at 7:17 PM, anfecora <anfecora@gmail.com> wrote:
Thank you Stoyan, i tried but i ended up creating a loop with the carrier, i believe this is more a asterisk receiving the package and ignoring the record-route and because i am just proxying the signalling it does ack to the contact, i have to find a way to tell asterisk that answer everything to kamailio and kamailio must respond to the carrier to the proper to header i am clueless here, now thinking to install rtpproxy  to achieve that, any other sugestions .
thanks.


U 2013/10/23 17:26:16.846067 3.1.1.1:5060 -> 1.1.1.2:5060
SIP/2.0 200 OK.
Session-Expires: 3600;refresher=uas.
Require: timer.
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000".
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=MSXB 4755 8544 IN IP4 3.1.1.2.
s=sip call.
c=IN IP4 204.15.40.111.
t=0 0.
m=audio 33408 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 2013/10/23 17:26:16.846201 1.1.1.2:5060 -> 1.1.1.1:5060
SIP/2.0 200 OK.
Session-Expires: 3600;refresher=uas.
Require: timer.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.
Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000".
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 202.
.
v=0.
o=MSXB 4755 8544 IN IP4 3.1.1.2.
s=sip call.
c=IN IP4 204.15.40.111.
t=0 0.
m=audio 33408 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


T 2013/10/23 17:26:16.846287 1.1.1.2:55305 -> 10.0.3.54:3306 [AP]
.....insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,kekuintid,type_call,dst_ip,carriercode,callmode ) values ('INVITE','as4bc322e9','3591552407-393967','7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060','200','OK','2013-10-23 17:26:16','sip:+19812457865@1.1.1.1','sip:23276341079@2.0.0.1','+19812457865','1.1.1.1','sip:76890723276341079@3.1.1.1:5060','sip:23276341079@2.0.0.1','OUT')

U 2013/10/23 17:26:16.847421 1.1.1.1:5060 -> 1.1.1.2:5060
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport.
Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Max-Forwards: 70.
From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9.
To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.8.15-cert2.
Content-Length: 0.
.


On Thu, Oct 24, 2013 at 12:59 PM, Stoyan Mihaylov <stoyan.v.mihaylov@gmail.com> wrote:
I had same problem - with BYE also.
My "go around" was (replaced name of domain and IP of kamailio):

route[ACKBYE] {
#!ifdef WITH_MYFORWARD
if(($sht(forw=>$ft))=~$td){
$du=$sht(forw=>$ft);
}else if((($td=="name.of.company.com")||($td=="ip"))&&($si=="ip")){
$du=$sht(forw=>$ft);
return;
}
#!endif
return;
}

route[PSTNINVITE] {
#!ifdef WITH_MYFORWARD
if(is_method("INVITE")){
ds_select_dst("1","4");
$sht(forw=>$ft)=$du;
sl_send_reply("100","Trying");
route(RELAY);
exit();
}
#!endif

return;
}

Meaning - during invite, I store du (to allow more then one Asterisk behind kamailio)
and on ACK or BYE - I check td and si. Not sure I am correct, but it works from long time, although load is not high.
PS
You will need to set in the beginning
modparam("htable", "htable", "forw=>size=8;autoexpire=7200;")

and you need to put routes in proper places.

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_______________________________________________
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