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.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
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.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
CSeq: 102 INVITE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE.
Contact: <sip:76890723276341079@3.1.1.2:5060>.
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
ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0.
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.
Contact: <sip:+19812457865@1.1.1.1:5060>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060.
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.mihay...@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.
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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