<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=Windows-1252">
<style type="text/css" style="display:none;"> P {margin-top:0;margin-bottom:0;} </style>
</head>
<body dir="ltr">
<div style="font-family: Calibri, Arial, Helvetica, sans-serif; font-size: 12pt; color: rgb(0, 0, 0); background-color: rgb(255, 255, 255);">
<span style="color: black; font-size: 12pt; background-color: white;">Hi again Ben,</span>
<div style="color: black; font-size: 12pt; background-color: white;"><br>
</div>
<div style="color: black; font-size: 12pt; background-color: white;">Not rewriting the Contact lets BYE work from the UAS, so that's fixed, many thanks. Now the only problem is the media from UAS to the UAC. The INVITE outbound looks like this now:</div>
<div style="color: black; font-size: 12pt; background-color: white;"><br>
</div>
<div style="color: black; font-size: 12pt; background-color: white;"><span></span>Session Initiation Protocol (INVITE)
<div> Request-Line: INVITE sip:5760@g---.spitfirevoiceapps.net:5060;transport=TCP SIP/2.0</div>
<div> Method: INVITE</div>
<div> Request-URI: sip:5760@g---.spitfirevoiceapps.net:5060;transport=TCP</div>
<div> [Resent Packet: False]</div>
<div> Message Header</div>
<div> Record-Route: <sip:195.196.197.198:5060;transport=tcp;r2=on;lr;ftag=pydzo;nat=yes></div>
<div> Record-Route: <sip:10.20.12.125;r2=on;lr;ftag=pydzo;nat=yes></div>
<div> Via: SIP/2.0/TCP 195.196.197.198:5060;branch=z9hG4bKee12.10e664c4.0</div>
<div> Via: SIP/2.0/UDP 10.193.169.56;received=10.193.169.56;rport=5060;branch=z9hG4bKycrgiemh</div>
<div> Max-Forwards: 69</div>
<div> To: <sip:+5760@195.196.197.198></div>
<div> From: "Alex" <sip:1838@195.196.197.198>;tag=pydzo</div>
<div> Call-ID: vfytldehdpikbnt@ryzing</div>
<div> [Generated Call-ID: vfytldehdpikbnt@ryzing]</div>
<div> CSeq: 224 INVITE</div>
<div> Contact: <sip:1838@10.193.169.56;transport=udp></div>
<div> Content-Type: application/sdp</div>
<div> Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE</div>
<div> Supported: replaces,norefersub,100rel</div>
<div> User-Agent: Twinkle/1.10.2</div>
<div> Content-Length: 339</div>
<div> Message Body</div>
<div> Session Description Protocol</div>
<div> Session Description Protocol Version (v): 0</div>
<div> Owner/Creator, Session Id (o): twinkle 1706657473 1887886456 IN IP4 10.193.169.56</div>
<div> Session Name (s): -</div>
<div> Connection Information (c): IN IP4 195.196.197.198</div>
<div> Time Description, active time (t): 0 0</div>
<div> Media Description, name and address (m): audio 35510 RTP/AVP 98 97 8 0 3 101</div>
<div> Media Attribute (a): rtpmap:98 speex/16000</div>
<div> Media Attribute (a): rtpmap:97 speex/8000</div>
<div> Media Attribute (a): rtpmap:8 PCMA/8000</div>
<div> Media Attribute (a): rtpmap:0 PCMU/8000</div>
<div> Media Attribute (a): rtpmap:3 GSM/8000</div>
<div> Media Attribute (a): rtpmap:101 telephone-event/8000</div>
<div> Media Attribute (a): fmtp:101 0-15</div>
<div> Media Attribute (a): sendrecv</div>
<div> Media Attribute (a): rtcp:35511</div>
<div> Media Attribute (a): ptime:20</div>
<span> [Generated Call-ID: vfytldehdpikbnt@ryzing]</span></div>
<div style="color: black; font-size: 12pt; background-color: white;"><span><br>
</span></div>
<div style="color: black; font-size: 12pt; background-color: white;"><span></span>But the audio coming back towards the UAC seems to be hitting the external interface of the proxy and not getting forwarded to the UAC on 10.193.169.56:</div>
<div style="color: black; font-size: 12pt; background-color: white;"><br>
</div>
<div style="color: black; font-size: 12pt; background-color: white;">Frame 21: 216 bytes on wire (1728 bits), 216 bytes captured (1728 bits)
<div>Linux cooked capture v1</div>
<div>Internet Protocol Version 4, Src: 62.63.64.65, Dst: 10.20.12.124</div>
<div> 0100 .... = Version: 4</div>
<div> .... 0101 = Header Length: 20 bytes (5)</div>
<div> Differentiated Services Field: 0x00 (DSCP: CS0, ECN: Not-ECT)</div>
<div> Total Length: 200</div>
<div> Identification: 0x848d (33933)</div>
<div> Flags: 0x40, Don't fragment</div>
<div> Fragment Offset: 0</div>
<div> Time to Live: 54</div>
<div> Protocol: UDP (17)</div>
<div> Header Checksum: 0x67f9 [validation disabled]</div>
<div> [Header checksum status: Unverified]</div>
<div> Source Address: 62.63.64.65</div>
<div> Destination Address: 10.20.12.124</div>
<div>User Datagram Protocol, Src Port: 10466, Dst Port: 35510</div>
<div> Source Port: 10466</div>
<div> Destination Port: 35510</div>
<div> Length: 180</div>
<div> Checksum: 0x9955 [unverified]</div>
<div> [Checksum Status: Unverified]</div>
<div> [Stream index: 4]</div>
<div> [Timestamps]</div>
<div> UDP payload (172 bytes)</div>
<div>Real-Time Transport Protocol</div>
<div> [Stream setup by SDP (frame 14)]</div>
<div> 10.. .... = Version: RFC 1889 Version (2)</div>
<div> ..0. .... = Padding: False</div>
<div> ...0 .... = Extension: False</div>
<div> .... 0000 = Contributing source identifiers count: 0</div>
<div> 0... .... = Marker: False</div>
<div> Payload type: ITU-T G.711 PCMA (8)</div>
<div> Sequence number: 25470</div>
<div> [Extended sequence number: 91006]</div>
<div> Timestamp: 1183222988</div>
<div> Synchronization Source identifier: 0x7485ecc6 (1954933958)</div>
<span> Payload: d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d5d45554d55454…</span></div>
<div style="color: black; font-size: 12pt; background-color: white;"><span><br>
</span></div>
<div style="color: black; font-size: 12pt; background-color: white;"><span>I'm running RPTEngine and it doesn't seem to be complaining - I've tried leaving the (c) address in SDP to the internal address but then UAC->UAS RTP goes straight to the 3CX, and sill
no UAS->UAC audio.<br>
</span></div>
<div style="color: black; font-size: 12pt; background-color: white;"><span><br>
</span></div>
<div style="color: black; font-size: 12pt; background-color: white;"><span>Wish we had IPv6 in this situation, would make everything work seamlessly!</span></div>
<div style="color: black; font-size: 12pt; background-color: white;"><span><br>
</span></div>
<div style="color: black; font-size: 12pt; background-color: white;"><span>Many thanks again for taking the time to help me.</span></div>
<div style="color: black; font-size: 12pt; background-color: white;"><span><br>
</span></div>
<div style="color: black; font-size: 12pt; background-color: white;"><span>Best regards</span></div>
<div style="color: black; font-size: 12pt; background-color: white;"><span><br>
</span></div>
<span style="color: black; font-size: 12pt; background-color: white;"><span>Alex</span></span><br>
</div>
<div id="appendonsend"></div>
<hr style="display:inline-block;width:98%" tabindex="-1">
<div id="divRplyFwdMsg" dir="ltr"><font face="Calibri, sans-serif" style="font-size:11pt" color="#000000"><b>From:</b> Users <users-bounces@lists.opensips.org> on behalf of Ben Newlin <Ben.Newlin@genesys.com><br>
<b>Sent:</b> 23 July 2021 16:14<br>
<b>To:</b> OpenSIPS users mailling list <users@lists.opensips.org><br>
<b>Subject:</b> Re: [OpenSIPS-Users] Double record-route trouble</font>
<div> </div>
</div>
<style>
<!--
@font-face
{font-family:"Cambria Math"}
@font-face
{font-family:Calibri}
p.x_MsoNormal, li.x_MsoNormal, div.x_MsoNormal
{margin:0in;
font-size:11.0pt;
font-family:"Calibri",sans-serif}
a:link, span.x_MsoHyperlink
{color:blue;
text-decoration:underline}
span.x_EmailStyle19
{font-family:"Calibri",sans-serif;
color:windowtext}
.x_MsoChpDefault
{font-size:10.0pt}
@page WordSection1
{margin:1.0in 1.0in 1.0in 1.0in}
div.x_WordSection1
{}
-->
</style>
<div lang="EN-US" link="blue" vlink="purple" style="word-wrap:break-word">
<div class="x_WordSection1">
<p class="x_MsoNormal">Alex,</p>
<p class="x_MsoNormal"> </p>
<p class="x_MsoNormal">Without seeing the INVITE I can’t be sure, but it seems like your problem is in your setting of the Contact header on the outbound INVITE to 3CX. The value of the Contact header is what dictates the RURI of sequential requests. The BYE
from 3CX is being sent to your OpenSIPS public interface, not to the Softphone IP. Because the RURI address matches the top Route header, OpenSIPS assumes that loose routing is not supported, which is per the RFC I believe, and that is why you see this behavior.
Replacing the URI with the contents of the route header is the old RFC 2543 routing mechanism.</p>
<p class="x_MsoNormal"> </p>
<p class="x_MsoNormal">If you are going to act as a proxy and use Record-Routing, you need to leave the Contact header URI pointing to the original UAC (the phone), not OpenSIPS. If you are going to change the Contact header to point to OpenSIPS, then Record-Routing
is not necessary. You should instead be using either Topology Hiding or B2BUA to handle that.</p>
<p class="x_MsoNormal"> </p>
<p class="x_MsoNormal"><span style="color:black">Ben Newlin </span></p>
<p class="x_MsoNormal"> </p>
<div style="border:none; border-top:solid #B5C4DF 1.0pt; padding:3.0pt 0in 0in 0in">
<p class="x_MsoNormal" style="margin-bottom:12.0pt"><b><span style="font-size:12.0pt; color:black">From:
</span></b><span style="font-size:12.0pt; color:black">Users <users-bounces@lists.opensips.org> on behalf of Alex Crow <alex.crow@geotogether.com><br>
<b>Date: </b>Friday, July 23, 2021 at 11:02 AM<br>
<b>To: </b>users@lists.opensips.org <users@lists.opensips.org><br>
<b>Subject: </b>[OpenSIPS-Users] Double record-route trouble</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">Hi all,</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">I am setting up a proxy to route calls between a cloud 3CX instance and MS Teams Direct Routing. At the moment I'm just trying to get the OpenSIPS to 3CX connection
working, testing with a softphone pointed at the proxy and attempting to make calls to extensions on the 3CX.</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">Opensips is behind NAT with two sockets configured, one TCP with an "as" setting for the public address of the NAT, one UDP which is where the softphone connects. Outbound
calls from the softphone as UAC to 3CX extensions as UAS ring, show established when answered, and I get audio at least one way (I think the one way audio may be a firewall issue). There are no registrations being used by any endpoint.</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">Opensips adds a double record-route header as expected for both the receiving and sending interface.</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">The problem I am having is that a BYE from one of the extensions on the 3CX (being a callee/UAS) arrives at OpenSIPs on the public-facing socket, with a double route
header, in the order of public, private, with lr and r2=on set, but loose_route() does not behave according to the RFCs. I believe it should remote both route headers and then send the BYE to the destination in the RURI, spiralling back into the proxy via
the sip address in the last Route: header and then being routed to the softphone/UAC.</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">Instead, it seems to replace the RURI with the sip address of the last Route header, and sends the message via TCP to the internal socket (which I therefore had to add
TCP to - if I dont have both UDP and TCP on the internal socket, we get a failed TCP connection here), thus causing a loop and an eventual Too Many Hops.</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">My openSIPS config is here:</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"><a href="https://pastebin.com/4t006vfn">https://pastebin.com/4t006vfn</a></span></p>
</div>
<p class="x_MsoNormal"> </p>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">A packet capture of the BYE and subsequent problem, the public facing private ip is 10.20.12.124 (TCP socket), advertised as 195.196.197.198, and the internal private
IP (facing the softphone) is 10.20.12.125 with a TCP and a UDP socket. 3CX is 62.63.64.65. Softphone (UAC) uses UDP, 3CX uses TCP. SIP Domain of the proxy and softphone is 195.196.197.198, the softphone's SIP username is 1838, I'm calling extension 5760 on
the 3CX.</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">BYE from 3XC (from UAS)</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">Session Initiation Protocol (BYE)
</span></p>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Request-Line: BYE sip:1838@195.196.197.198:5060;transport=tcp SIP/2.0</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Message Header</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Via: SIP/2.0/TCP 62.63.64.65:5060;branch=z9hG4bK-524287-1---77636e709985902e;rport</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Max-Forwards: 70</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Route: <sip:195.196.197.198:5060;transport=tcp;lr;r2=on;ftag=lihnv;nat=yes></span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Route: <sip:10.20.12.125;r2=on;lr;ftag=lihnv;nat=yes></span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Contact: <sip:5760@62.63.64.65:5060;transport=TCP></span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> To: "Alex" <sip:1838@195.196.197.198>;tag=lihnv</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> From: <sip:+5760@195.196.197.198>;tag=f6ce187d</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Call-ID: mpytwgabkcpnkef@ryzing</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> [Generated Call-ID: mpytwgabkcpnkef@ryzing]</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> CSeq: 2 BYE</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> User-Agent: 3CXPhoneSystem 16.0.8.9 (9)</span></p>
</div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Content-Length: 0</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">BYE after loose_route():</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">Session Initiation Protocol (BYE)
</span></p>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Request-Line: BYE sip:10.20.12.125;r2=on;lr;ftag=lihnv;nat=yes SIP/2.0</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Message Header</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Via: SIP/2.0/TCP 195.196.197.198:5060;branch=z9hG4bKcd21.3682afa6.0;i=a10f2936</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Via: SIP/2.0/TCP 62.63.64.65:5060;received=62.63.64.65;branch=z9hG4bK-524287-1---77636e709985902e;rport=44953</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Max-Forwards: 69</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Contact: <sip:5760@62.63.64.65:5060;transport=TCP></span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> To: "Alex" <sip:1838@195.196.197.198>;tag=lihnv</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> From: <sip:+5760@195.196.197.198>;tag=f6ce187d</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Call-ID: mpytwgabkcpnkef@ryzing</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> [Generated Call-ID: mpytwgabkcpnkef@ryzing]</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> CSeq: 2 BYE</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> User-Agent: 3CXPhoneSystem 16.0.8.9 (9)</span></p>
</div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Content-Length: 0</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">Then this just loops around as expected due to the RURI in the above message:</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">Session Initiation Protocol (BYE)
</span></p>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Request-Line: BYE sip:10.20.12.125;r2=on;lr;ftag=lihnv;nat=yes SIP/2.0</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Message Header</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Via: SIP/2.0/UDP 10.20.12.125:5060;branch=z9hG4bKcd21.4682afa6.0;i=c10f2936</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Via: SIP/2.0/TCP 195.196.197.198:5060;rport=47455;received=10.20.12.124;branch=z9hG4bKcd21.3682afa6.0;i=a10f2936</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Via: SIP/2.0/TCP 62.63.64.65:5060;received=62.63.64.65;branch=z9hG4bK-524287-1---77636e709985902e;rport=44953</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Max-Forwards: 68</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Contact: <sip:5760@62.63.64.65:5060;transport=TCP></span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> To: "Alex" <sip:1838@195.196.197.198>;tag=lihnv</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> From: <sip:+5760@195.196.197.198>;tag=f6ce187d</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Call-ID: mpytwgabkcpnkef@ryzing</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> [Generated Call-ID: mpytwgabkcpnkef@ryzing]</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> CSeq: 2 BYE</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> User-Agent: 3CXPhoneSystem 16.0.8.9 (9)</span></p>
</div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> Content-Length: 0</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">Any clues or ways of resolving this would be very much appreciated!</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">Best regards</span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black"> </span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12.0pt; color:black">Alex</span></p>
</div>
<p class="x_MsoNormal">DISCLAIMER: This email and any attachments are sent in confidence, subject to applicable legal privilege and upon the basis that the recipient will conduct appropriate checks. If you receive this email in error, please telephone us upon
receipt: you are strictly prohibited from using, copying or disseminating it or any information contained in it save to the intended recipient. Internet communications are not secure and Green Energy Options Ltd is not responsible for their abuse by third
parties, nor for any alteration or corruption in transmission, nor for any damage or loss caused by any virus or other defect. Green Energy Options Limited. Registered office: 3 St. Mary's Court, Main Street, Hardwick, Cambridge CB23 7QS Registered in England
no. 5783558 </p>
</div>
</div>
DISCLAIMER: This email and any attachments are sent in confidence, subject to applicable legal privilege and upon the basis that the recipient will conduct appropriate checks. If you receive this email in error, please telephone us upon receipt: you are strictly
prohibited from using, copying or disseminating it or any information contained in it save to the intended recipient. Internet communications are not secure and Green Energy Options Ltd is not responsible for their abuse by third parties, nor for any alteration
or corruption in transmission, nor for any damage or loss caused by any virus or other defect. Green Energy Options Limited. Registered office: 3 St. Mary's Court, Main Street, Hardwick, Cambridge CB23 7QS Registered in England no. 5783558
</body>
</html>