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<tt>Hi John,<br>
<br>
Probably the 5061 port (specific to TLS) is not re-written during
the routing process - for example see the 5061 in the REGISTER
RURI. How do you route the REGISTER from OpenSIPS to Asterisk ?
Dispatcher ? or ?<br>
<br>
Regards, <br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="https://www.opensips-solutions.com">https://www.opensips-solutions.com</a>
OpenSIPS Summit 2019
<a class="moz-txt-link-freetext" href="https://www.opensips.org/events/Summit-2019Amsterdam/">https://www.opensips.org/events/Summit-2019Amsterdam/</a>
</pre>
<div class="moz-cite-prefix">On 03/15/2019 01:03 AM, John Kiniston
wrote:<br>
</div>
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cite="mid:CAFJQOGek1LZExgijOiaWdQ3a+rXTXmA95Y62aRAur3WcY96VSA@mail.gmail.com">
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<div>I have a proxy that works fine with UDP handling
mid-registrar and mediaproxy duties in front of my
Asterisk Servers.<br>
<br>
</div>
I'd like to enable TLS on my proxy and switch my
phones to registering via TLS.<br>
<br>
</div>
I've got my TLS keys working now and I see my phone
sending requests to OpenSIPS where they are being
processed.<br>
<br>
</div>
However when OpenSIPS is sending the traffic to my
Asterisk boxes it's sending SIP traffic to port 5061
instead of sending TLS.<br>
<br>
U 2019/03/14 15:54:33.985636 <Proxy>:5060 ->
<Asterisk>:5061 #32<br>
REGISTER sip:<pbx>:5061 SIP/2.0.<br>
Via: SIP/2.0/UDP
Proxy:5060;branch=z9hG4bKffdc.94c66575.0;i=a64b8973.<br>
Via: SIP/2.0/TLS
172.16.52.69:12332;received=64.119.40.170;branch=z9hG4bK3157565843;rport=12332.<br>
From: "Test"
<<a class="moz-txt-link-freetext" href="sip:7004@">sip:7004@</a><pbx>:5061>;tag=1087298031.<br>
To: "Test" <<a class="moz-txt-link-freetext" href="sip:7004@">sip:7004@</a><pbx>:5061>.<br>
Call-ID: <a href="mailto:0_425741679@172.16.52.69"
moz-do-not-send="true">0_425741679@172.16.52.69</a>.<br>
CSeq: 1 REGISTER.<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:7004@Proxy:5060"><sip:7004@Proxy:5060></a>.<br>
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS,
NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE,
MESSAGE.<br>
Max-Forwards: 69.<br>
User-Agent: Yealink SIP-T42G 29.83.0.50<br>
Expires: 3600.<br>
Allow-Events: talk,hold,conference,refer,check-sync.<br>
Mac: 00:15:65:72:e5:f3.<br>
Line: 1.<br>
Content-Length: 0.<br>
<br>
</div>
<div>I believe this is because the initial request opensips
is recieving from the phone has 'sip' in the register?<br>
<br>
REGISTER sip:<pbx>:5061 SIP/2.0 <br>
Via: SIP/2.0/TLS
172.16.52.69:12122;branch=z9hG4bK2625059359;rport <br>
From: "Test" <a class="moz-txt-link-rfc2396E" href="sip:7004@pbx:5061"><sip:7004@pbx:5061></a>;tag=1299551992
<br>
To: "Test" <a class="moz-txt-link-rfc2396E" href="sip:7004@pbx:5061"><sip:7004@pbx:5061></a> <br>
Call-ID: <a href="mailto:0_292898054@172.16.52.69"
moz-do-not-send="true">0_292898054@172.16.52.69</a> <br>
CSeq: 1 REGISTER <br>
Contact:
<a class="moz-txt-link-rfc2396E" href="sip:7004@172.16.52.69:12122;transport=TLS"><sip:7004@172.16.52.69:12122;transport=TLS></a> <br>
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS,
NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE,
MESSAGE <br>
Max-Forwards: 70 <br>
User-Agent: Yealink SIP-T42G 29.83.0.50<br>
Expires: 3600 <br>
Allow-Events: talk,hold,conference,refer,check-sync <br>
Mac: 00:15:65:72:e5:f3 <br>
Line: 1 <br>
Content-Length: 0 <br>
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<div>What can I do here? I've tried playing with
rewriting $du and $fs but that doesn't seem to
be working.<br>
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<div>-- <br>
<div dir="ltr" class="gmail_signature">A human
being should be able to change a diaper,
plan an invasion, butcher a hog, conn a
ship, design a building, write a sonnet,
balance accounts, build a wall, set a bone,
comfort the dying, take orders, give orders,
cooperate, act alone, solve equations,
analyze a new problem, pitch manure, program
a computer, cook a tasty meal, fight
efficiently, die gallantly. Specialization
is for insects.<br>
---Heinlein</div>
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