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<p>Sorry for delay on my reply..</p>
<p><br>
</p>
<p>I need to expalin better the situazione..</p>
<p>Customer1 Ip : 1.1.1.1<br>
Kamailio1 ip : 2.2.2.2<br>
Kamailio2 ip: 3.3.3.3<br>
CiscoGW ip: 4.4.4.4<br>
</p>
<p>Kamailio1 is on USA for example<br>
Kamailio2 is on Germany for example<br>
</p>
<p>Customer1 --> Kamailio platform1 --> Kamailio Platform2
--> CISCO GW SIP/TDM for PTSN termination<br>
</p>
<p>Customer1 is sending a call using his specific color 9999 to
number 4912345678 and from sender 151512345678</p>
<p>U 2016/08/10 09:54:29.250974 1.1.1.1:5060 ->2.2.2.2:5060<br>
INVITE sip:<b>9999</b><a class="moz-txt-link-abbreviated" href="mailto:4912345678@2.2.2.2">4912345678@2.2.2.2</a> SIP/2.0.<br>
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK06b62a40;rport.<br>
Max-Forwards: 70.<br>
From: "151512345678"
<a class="moz-txt-link-rfc2396E" href="sip:151512345678@1.1.1.1"><sip:151512345678@1.1.1.1></a>;tag=as7f0dee78.<br>
To: <sip:<b>9999</b><a class="moz-txt-link-abbreviated" href="mailto:4912345678@2.2.2.2">4912345678@2.2.2.2</a>>.<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:151512345678@1.1.1.1:5060"><sip:151512345678@1.1.1.1:5060></a>.<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060">406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060</a>.<br>
CSeq: 102 INVITE.<br>
User-Agent: Asterisk PBX 1.8.32.3.<br>
Date: Wed, 10 Aug 2016 08:54:27 GMT.<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLISH, MESSAGE.<br>
Supported: replaces, timer.<br>
Content-Type: application/sdp.<br>
Content-Length: 309.<br>
.<br>
v=0.<br>
o=root 869935480 869935480 IN IP4 1.1.1.1.<br>
s=Asterisk PBX 1.8.32.3.<br>
c=IN IP4 1.1.1.1.<br>
t=0 0.<br>
m=audio 15710 RTP/AVP 3 18 8 101.<br>
a=rtpmap:3 GSM/8000.<br>
a=rtpmap:18 G729/8000.<br>
a=fmtp:18 annexb=no.<br>
a=rtpmap:8 PCMA/8000.<br>
a=rtpmap:101 telephone-event/8000.<br>
a=fmtp:101 0-16.<br>
a=ptime:20.<br>
a=sendrecv.<br>
</p>
<p><br>
</p>
<p>After that the Kamailio1 platform is checking the LCR and route
it with the color of its supplier (9053) to Kamailio2. Kamailio2
is a supplier of Kamailio1</p>
<p>U 2016/08/10 09:54:29.2525272.2.2.2:5060 -> 3.3.3.3:5060<br>
INVITE sip:<b>9053</b><a class="moz-txt-link-abbreviated" href="mailto:4912345678@3.3.3.3">4912345678@3.3.3.3</a> SIP/2.0.<br>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:2.2.2.2;lr;did=4f3.8501;nat=yes"><sip:2.2.2.2;lr;did=4f3.8501;nat=yes></a>.<br>
Via:
SIP/2.0/UDP2.2.2.2:5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0.<br>
Via: SIP/2.0/UDP
1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060.<br>
Max-Forwards: 69.<br>
From: "151512345678"
<a class="moz-txt-link-rfc2396E" href="sip:151512345678@1.1.1.1"><sip:151512345678@1.1.1.1></a>;tag=as7f0dee78.<br>
To: <sip:<b>9053</b><a class="moz-txt-link-abbreviated" href="mailto:4912345678@3.3.3.3">4912345678@3.3.3.3</a>>.<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:151512345678@1.1.1.1:5060"><sip:151512345678@1.1.1.1:5060></a>.<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060">406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060</a>.<br>
CSeq: 102 INVITE.<br>
Date: Wed, 10 Aug 2016 08:54:27 GMT.<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLISH, MESSAGE.<br>
Supported: replaces, timer.<br>
Content-Type: application/sdp.<br>
Content-Length: 308.<br>
User-Agent: Fagians VOIP 2.4.<br>
.<br>
v=0.<br>
o=root 869935480 869935480 IN IP4 1.1.1.1.<br>
s=Asterisk PBX 1.8.32.3.<br>
c=IN IP4 51.254.158.37.<br>
t=0 0.<br>
m=audio 36398 RTP/AVP 3 18 8 101.<br>
a=rtpmap:3 GSM/8000.<br>
a=rtpmap:18 G729/8000.<br>
a=fmtp:18 annexb=no.<br>
a=rtpmap:8 PCMA/8000.<br>
a=rtpmap:101 telephone-event/8000.<br>
a=fmtp:101 0-16.<br>
a=ptime:20.<br>
a=sendrecv.<br>
<br>
</p>
<p>Kamailio2 use its LCR and send the call to Cisco Gateway that use
its color and send the call on termination to TDM Switch.<br>
Naturally Kamailio2 receive the replies from Cisco and send it
back to Kamailio1.</p>
<p><br>
</p>
<p>Here is the Session progress Kamailio1 receive from Kamailio2
that it got from Cisco.<br>
</p>
<p>U 2016/08/10 09:54:29.375669 3.3.3.3:5060 ->2.2.2.2:5060<br>
SIP/2.0 183 Session Progress.<br>
Via:
SIP/2.0/UDP2.2.2.2:5060;rport=5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0,SIP/2.0/UDP
1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060.<br>
From: "151512345678"
<a class="moz-txt-link-rfc2396E" href="sip:151512345678@1.1.1.1"><sip:151512345678@1.1.1.1></a>;tag=as7f0dee78.<br>
To: <sip:<b>9990</b><a class="moz-txt-link-abbreviated" href="mailto:4912345678@4.4.4.4">4912345678@4.4.4.4</a>>;tag=5F0E7DF4-172F.<br>
Date: Wed, 10 Aug 2016 08:54:29 GMT.<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060">406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060</a>.<br>
CSeq: 102 INVITE.<br>
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER.<br>
Allow-Events: telephone-event.<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:99904912345678@4.4.4.4:5060"><sip:99904912345678@4.4.4.4:5060></a>.<br>
Record-Route:
<a class="moz-txt-link-rfc2396E" href="sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes"><sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes></a>,<a class="moz-txt-link-rfc2396E" href="sip:2.2.2.2;lr;did=4f3.8501;nat=yes"><sip:2.2.2.2;lr;did=4f3.8501;nat=yes></a>.<br>
Content-Disposition: session;handling=required.<br>
Content-Type: application/sdp.<br>
Content-Length: 251.<br>
User-Agent: Fagians VOIP 2.4.<br>
.<br>
v=0.<br>
o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4.<br>
s=SIP Call.<br>
c=IN IP4 83.147.127.247.<br>
t=0 0.<br>
m=audio 58240 RTP/AVP 3 101.<br>
c=IN IP4 83.147.127.247.<br>
a=rtpmap:3 GSM/8000.<br>
a=rtpmap:101 telephone-event/8000.<br>
a=fmtp:101 0-16.<br>
a=ptime:10.<br>
<br>
</p>
<p>To: <a class="moz-txt-link-rfc2396E" href="sip:99904912345678@4.4.4.4"><sip:99904912345678@4.4.4.4></a>;tag=5F0E7DF4-172F.
->> 9990 is the color that use CISCO to terminate the call
on TDM Switch</p>
<p>After some other messages Kamailio1 receive the 200 OK and send
it back to Customer1</p>
<p><br>
</p>
<p>Kamailio2 --> Kamailio1<br>
</p>
<p>U 2016/08/10 09:54:39.507885 3.3.3.3:5060 ->2.2.2.2:5060<br>
SIP/2.0 200 OK.<br>
Via:
SIP/2.0/UDP2.2.2.2:5060;rport=5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0,SIP/2.0/UDP
1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060.<br>
From: "151512345678"
<a class="moz-txt-link-rfc2396E" href="sip:151512345678@1.1.1.1"><sip:151512345678@1.1.1.1></a>;tag=as7f0dee78.<br>
To: <sip:<b>9990</b><a class="moz-txt-link-abbreviated" href="mailto:4912345678@4.4.4.4">4912345678@4.4.4.4</a>>;tag=5F0E7DF4-172F.<br>
Date: Wed, 10 Aug 2016 08:54:29 GMT.<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060">406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060</a>.<br>
CSeq: 102 INVITE.<br>
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER.<br>
Supported: replaces.<br>
Allow-Events: telephone-event.<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:99904912345678@4.4.4.4:5060"><sip:99904912345678@4.4.4.4:5060></a>.<br>
Record-Route:
<a class="moz-txt-link-rfc2396E" href="sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes"><sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes></a>,<a class="moz-txt-link-rfc2396E" href="sip:2.2.2.2;lr;did=4f3.8501;nat=yes"><sip:2.2.2.2;lr;did=4f3.8501;nat=yes></a>.<br>
Content-Type: application/sdp.<br>
Content-Length: 251.<br>
User-Agent: Fagians VOIP 2.4.<br>
.<br>
v=0.<br>
o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4.<br>
s=SIP Call.<br>
c=IN IP4 83.147.127.247.<br>
t=0 0.<br>
m=audio 58240 RTP/AVP 3 101.<br>
c=IN IP4 83.147.127.247.<br>
a=rtpmap:3 GSM/8000.<br>
a=rtpmap:101 telephone-event/8000.<br>
a=fmtp:101 0-16.<br>
a=ptime:10.<br>
<br>
</p>
<p>Kamailio1 --> Customer1</p>
<p>U 2016/08/10 09:54:39.5120362.2.2.2:5060 -> 1.1.1.1:5060<br>
SIP/2.0 200 OK.<br>
Via: SIP/2.0/UDP
1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060.<br>
From: "151512345678"
<a class="moz-txt-link-rfc2396E" href="sip:151512345678@1.1.1.1"><sip:151512345678@1.1.1.1></a>;tag=as7f0dee78.<br>
To: <sip:<b>9990</b><a class="moz-txt-link-abbreviated" href="mailto:4912345678@4.4.4.4">4912345678@4.4.4.4</a>>;tag=5F0E7DF4-172F.<br>
Date: Wed, 10 Aug 2016 08:54:29 GMT.<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060">406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060</a>.<br>
CSeq: 102 INVITE.<br>
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER.<br>
Supported: replaces.<br>
Allow-Events: telephone-event.<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:99904912345678@4.4.4.4:5060"><sip:99904912345678@4.4.4.4:5060></a>.<br>
Record-Route:
<a class="moz-txt-link-rfc2396E" href="sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes"><sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes></a>,<a class="moz-txt-link-rfc2396E" href="sip:2.2.2.2;lr;did=4f3.8501;nat=yes"><sip:2.2.2.2;lr;did=4f3.8501;nat=yes></a>.<br>
Content-Type: application/sdp.<br>
Content-Length: 249.<br>
User-Agent: Fagians VOIP 2.4.<br>
.<br>
v=0.<br>
o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4.<br>
s=SIP Call.<br>
c=IN IP4 51.254.158.37.<br>
t=0 0.<br>
m=audio 56710 RTP/AVP 3 101.<br>
c=IN IP4 51.254.158.37.<br>
a=rtpmap:3 GSM/8000.<br>
a=rtpmap:101 telephone-event/8000.<br>
a=fmtp:101 0-16.<br>
a=ptime:10.<br>
<br>
</p>
<p>So the real question is how to fix that on Kamailio ?..</p>
<p>We need to use always the original messages and data into sdp
header when we talk with other parts..</p>
<p>On our configuration we permit to transit that modified
messages.. like you can see Customer1 is getting back datas
modified from CiscoGW.</p>
<p><br>
</p>
<p>Hope that will be more clear to you all..</p>
<p><br>
</p>
<p>Anyone can suggest us a way ?</p>
<p><br>
</p>
<p>Regards</p>
<p>Laura<br>
</p>
<br>
<div class="moz-cite-prefix">Il 01/08/16 14:25, Carsten Bock ha
scritto:<br>
</div>
<blockquote
cite="mid:CAOCjumFXuQyB4dST-p9jDWdGG67yyPmf_Q1AZvZR_r9Y9+dCnw@mail.gmail.com"
type="cite">
<div dir="ltr">Hi,
<div><br>
</div>
<div>do you use "uac_replace_from" or "uac_replace_to" in your
logic?</div>
<div><br>
</div>
<div>If not, it seems to me, that your supplier is messing
around with the SIP-Replies.</div>
<div><br>
Thanks,</div>
<div>Carsten</div>
<img moz-do-not-send="true"
src="mailbox:///Volumes/FAGIANO/Mail/ats.it/Drafts?number=6&si=6230090009280512&pi=6c904f76-3bd9-4b05-e4ff-1974d70d6b00"
style="display:none!important" height="1" width="1"></div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2016-08-01 14:10 GMT+02:00 Laura <span
dir="ltr"><<a moz-do-not-send="true"
href="mailto:red_dra@plugit.net" target="_blank">red_dra@plugit.net</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">Dear list,<br>
<br>
i'm asking here a question about Kamailio config.<br>
<br>
We are testing a wide area configuration of Kamailio over
separates<br>
countries and we are still facing with an issue.<br>
<br>
We configured Kamailio 4.3.5 with dialog support over the TM
modules and<br>
we use LCR module for menage ours LCRs rule set profiles.<br>
<br>
For some technicals reasons we use tech prefix for our
customer so for<br>
exaples customer1 send traffic to us with 1111 prefix,
customer2 send<br>
traffic to us with 2222 and something similar..<br>
<br>
Our supplier, of course, are using tech prefix too so for
examples if i<br>
want to send the call to supplier1 i need to use tech prefix
1789 or<br>
something similar..<br>
<br>
The point is..<br>
<br>
<br>
When customer1 is sending an invite to us.. it send us
something like<br>
(Bangladesh mobile 8801xxx)<br>
<br>
INVITE <a class="moz-txt-link-freetext" href="sip:11118801xxxxxxx@aaa.bbb.ccc.ddd">sip:11118801xxxxxxx@aaa.bbb.ccc.ddd</a><br>
<br>
Our Kamailio will reply with the Trying and then it goes to
LCR module<br>
and match our supplier1 so it make a new invite like this<br>
<br>
INVITE <a class="moz-txt-link-freetext" href="sip:17898801xxxxxx@supplier.ip">sip:17898801xxxxxx@supplier.ip</a><br>
<br>
The problem come when supplier1 reply to us and we replies
back to<br>
customer1..<br>
<br>
Customer1 view the From: field with the 17898801xxxxxx
numbers.. and<br>
some of our customers don't like it.<br>
<br>
We don't use anymore the topoh module becuase we found some
troubles<br>
using it.. so..<br>
<br>
Is there a way that we can use for fix this situation ?<br>
<br>
<br>
Best regards.<br>
<br>
<br>
<br>
_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing list<br>
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href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
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href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users"
rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
</blockquote>
</div>
<br>
<br clear="all">
<div><br>
</div>
-- <br>
<div class="gmail_signature" data-smartmail="gmail_signature">Carsten
Bock<br>
CEO (Geschäftsführer)<br>
<br>
ng-voice GmbH<br>
Millerntorplatz 1<br>
20359 Hamburg / Germany<br>
<br>
<a moz-do-not-send="true" href="http://www.ng-voice.com"
target="_blank">http://www.ng-voice.com</a><br>
mailto:<a moz-do-not-send="true"
href="mailto:carsten@ng-voice.com" target="_blank">carsten@ng-voice.com</a><br>
<br>
Office +49 40 5247593-40<br>
Fax +49 40 5247593-99<br>
<br>
Sitz der Gesellschaft: Hamburg<br>
Registergericht: Amtsgericht Hamburg, HRB 120189<br>
Geschäftsführer: Carsten Bock<br>
Ust-ID: DE279344284<br>
<br>
Hier finden Sie unsere handelsrechtlichen Pflichtangaben:<br>
<a moz-do-not-send="true"
href="http://www.ng-voice.com/imprint/" target="_blank">http://www.ng-voice.com/imprint/</a></div>
</div>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
</blockquote>
<br>
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