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<div class="moz-cite-prefix">Hi,<br>
general speaking you have the following scenarios:<br>
<br>
== Call between subscribers ==<br>
Subscriber A call B, both are under the same Domain<br>
<br>
1. A send INVITE to SPCE<br>
2. INBOUND REWRITE RULES of A's Domain is used<br>
3. SPCE lookup B as callee<br>
4. OUTBOUND REWRITE RULES of B's Domain is used<br>
5. SPCE send INVITE to B<br>
<br>
<br>
== Call from subscribers to PEER ==<br>
Subscriber A call outbound.<br>
<br>
1. A send INVITE to SPCE<br>
2. INBOUND REWRITE RULES of A's Domain is used<br>
3. SPCE Select the peer based on Caller/Callee<br>
4. OUTBOUND REWRITE RULES of PEER is used<br>
5. SPCE send INVITE to the peer<br>
<br>
<br>
== Call from peer ==<br>
Incoming call from peer to Subscriber A<br>
<br>
1. PEER send INVITE to SPCE<br>
2. INBOUND REWRITE RULES of PEER is used<br>
3. SPCE lookup A as callee<br>
4. OUTBOUND REWRITE RULES of A's Domain is used<br>
5. SPCE send INVITE to A<br>
<br>
<br>
<br>
NOTE: You can set Rewrite rules set in DOMAIN level. If you set
another Rewrite rules set on SUBSCRIBER's preferences level, this
will overwrite the Domain set.<br>
<br>
<br>
Regards,<br>
Daniel<br>
<br>
<br>
<br>
<br>
On 12/30/2014 02:30 PM, <a class="moz-txt-link-abbreviated" href="mailto:mig@gmx.ch">mig@gmx.ch</a> wrote:<br>
</div>
<blockquote
cite="mid:trinity-9eed117a-9897-4038-99d3-10a06f0d8709-1419946250214@3capp-gmx-bs01"
type="cite">
<div style="font-family: Verdana;font-size: 12.0px;">
<div>
<div>Hi all,</div>
<div>First of all sorry for my stupid questions but I'm a
newbie and installed a couple of days/weeks SIP:Provider CE
Version mr3.6.2 only for test purpose in my LAB.<br>
On chapter 6.6 Configuring Rewrite Rule Sets (Handbook) it
is written that on the NGCP every number is treated in E.164
format. Does that mean with every phone number From/To
(caller/callee) ?<br>
I'm in switzerland and have a DDI 0435444390 - 99 assigned
to me. I receive from the provider all the number with
leading 0, that means for national 0 and international 00.
The provider expects to receive the whole number from my
DDI, equal 0435444390 (10 digits) and if it is a
international call the callee number with 00.</div>
<div> </div>
<div> </div>
<div>How are the Rewrite Rule Sets processed? I mean they can
be assigned either to a peering,domain or subscriber. If a
call is passing a peering server which has a rewrite rule
set assigned I guess this one is processed, but what if the
domain has also one set? are both of them proccessed or only
that one from the peering server?<br>
How can I troubleshoot the rewrite rules what has been
processed on a call so that I can check if the regex works
as desired?</div>
<div> </div>
<div> </div>
<div>Example of an in and outgoing call:</div>
<div> </div>
<div>Swiss numbering plan:
<a class="moz-txt-link-freetext" href="http://www.bakom.admin.ch/themen/telekom/00479/00604/index.html?lang=en&download=NHzLpZeg7t,lnp6I0NTU042l2Z6ln1ad1IZn4Z2qZpnO2Yuq2Z6gpJCDdIN5f2ym162epYbg2c_JjKbNoKSn6A">http://www.bakom.admin.ch/themen/telekom/00479/00604/index.html?lang=en&download=NHzLpZeg7t,lnp6I0NTU042l2Z6ln1ad1IZn4Z2qZpnO2Yuq2Z6gpJCDdIN5f2ym162epYbg2c_JjKbNoKSn6A</a>--</div>
<div><br>
<cc> = 41<br>
<ac> = 43<br>
<sn> = 5444390 - 99</div>
<div><IP Phone> IP address 172.16.1.14<br>
<NGCP> IP address 172.16.1.7<br>
<PSTN> IP address 62.2.46.12</div>
<div> </div>
<div>originating call <IP Phone> , terminating
<PSTN><br>
<IP Phone> ---> <NGCP> --->
<Peering> ---> <PSTN></div>
<div> </div>
<div>INVITE <a class="moz-txt-link-abbreviated" href="mailto:sip:0800800800@sip.migmig.lan">sip:0800800800@sip.migmig.lan</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.1.14:5060;branch=z9hG4bK-8d69118c<br>
From: "0435444395"
<a class="moz-txt-link-rfc2396E" href="mailto:sip:0435444395@sip.migmig.lan"><sip:0435444395@sip.migmig.lan></a>;tag=c553358ee74d6b93o2<br>
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:0800800800@sip.migmig.lan"><sip:0800800800@sip.migmig.lan></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:c821dd6a-2a6d8567@172.16.1.14">c821dd6a-2a6d8567@172.16.1.14</a><br>
CSeq: 101 INVITE<br>
Max-Forwards: 70<br>
Contact: "0435444395"
<a class="moz-txt-link-rfc2396E" href="mailto:sip:0435444395@172.16.1.14:5060"><sip:0435444395@172.16.1.14:5060></a><br>
Expires: 240<br>
User-Agent: Linksys/SPA941-5.1.8<br>
Content-Length: 208<br>
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS,
REFER<br>
Supported: replaces<br>
Content-Type: application/sdp</div>
<div>v=0<br>
o=- 22782282 22782282 IN IP4 172.16.1.14<br>
s=-<br>
c=IN IP4 172.16.1.14<br>
t=0 0<br>
m=audio 16462 RTP/AVP 8 101<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=ptime:20<br>
a=sendrecv</div>
<div> </div>
<div> </div>
<div> </div>
<div>originating call <PSTN> , terminating <IP
Phone><br>
<PSTN> ---> <Peering> ---> <NGCP>
---> <IP Phone></div>
<div> </div>
<div>INVITE <a class="moz-txt-link-abbreviated" href="mailto:sip:0435444395@172.16.1.7:5060;user=phone">sip:0435444395@172.16.1.7:5060;user=phone</a> SIP/2.0<br>
Via: SIP/2.0/UDP
62.2.46.12:5060;branch=z9hG4bK18f4ok00eg51m0n9j0m0.1<br>
From: "0445777593"
<a class="moz-txt-link-rfc2396E" href="mailto:sip:0445777593@212.55.198.150;user=phone"><sip:0445777593@212.55.198.150;user=phone></a>;tag=528762297<br>
To: 0435444395
<a class="moz-txt-link-rfc2396E" href="mailto:sip:0435444395@vbcOTF005.cablecom.net:5060;user=phone"><sip:0435444395@vbcOTF005.cablecom.net:5060;user=phone></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:4aa9fda56b9308d037bd213252ff18d4@212.55.198.150">4aa9fda56b9308d037bd213252ff18d4@212.55.198.150</a><br>
CSeq: 1 INVITE<br>
Max-Forwards: 67<br>
Supported: timer<br>
Session-Expires: 1800<br>
Min-SE: 1800<br>
Contact:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:0445777593@62.2.46.12:5060;transport=udp"><sip:0445777593@62.2.46.12:5060;transport=udp></a><br>
Allow:
INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE<br>
Date: Sun, 21 Dec 2014 20:54:08 GMT<br>
Content-Type: application/sdp<br>
Content-Length: 260</div>
<div>v=0<br>
o=root 1858387754 1858387754 IN IP4 62.2.46.12<br>
s=Asterisk PBX 1.6.1.24<br>
c=IN IP4 62.2.46.12<br>
t=0 0<br>
m=audio 16626 RTP/AVP 8 101<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=sendrecv<br>
a=ptime:20</div>
<div><br>
Regards,</div>
<div>Miguel</div>
</div>
</div>
<br>
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<br>
<pre wrap="">_______________________________________________
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</blockquote>
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