<div dir="auto">Awesome, thanks! It working perfectly. </div><div class="gmail_extra"><br><div class="gmail_quote">On 23/03/2018 12:33 AM, "Andrew Pogrebennyk" <<a href="mailto:apogrebennyk@sipwise.com">apogrebennyk@sipwise.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div class="m_7988023869890242840moz-cite-prefix">Hi Chris,<br>
you should try editing the files:<br>
/etc/ngcp-config/templates/<wbr>etc/kamailio/lb/kamailio.cfg.<wbr>customtt.tt2<br>
/etc/ngcp-config/templates/<wbr>etc/kamailio/proxy/kamailio.<wbr>cfg.customtt.tt2<br>
(create them from tt2 according to handbook if they don't exit)<br>
and add the line<br>
modparam("rr", "<span class="m_7988023869890242840il">enable_full_lr</span>", 0)<br>
in both. This should do the trick!<br>
<br>
Regards,<br>
Andrew<br>
<br>
On 03/22/2018 08:51 AM, Chris Hoffmann wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">Hi,
<div><br>
</div>
<div>Over the last few weeks I have been experimenting with
SipWise. I have attempted to set up lync as a subscriber and
can make calls with 2 way audio however Lync does not appear
to be reciving the SIP200 message as the interface still shows
ringing. I have done a number of wireshark captures from the
Lync server and compared calls via Asterisk which work and
calls via SipWise which have the above issue. </div>
<div><br>
</div>
<div><br>
</div>
<div>
<div>SIP 200 that doesn't get recognised by Lync</div>
<div>------------------------------<wbr>---------------</div>
<div>SIP/2.0 200 OK</div>
<div>Record-Route:
<a class="m_7988023869890242840moz-txt-link-rfc2396E"><sip:127.0.0.1:5062;lr=on;<wbr>ftag=6210295535;did=1c9.d452;<wbr>ice_caller=strip;ice_callee=<wbr>strip;aset=50;rtpprx=yes;vsf=<wbr>anl6aVVKfUJ5RXlpRmYJGDtnIxIzMy<wbr>x2HhVwEzN6DR4kF0UZBlsNMQ--></a></div>
<div>Record-Route:
<<a class="m_7988023869890242840moz-txt-link-freetext">sip:127.0.0.1;r2=on;lr=on;<wbr>ftag=6210295535;ngcplb=yes;<wbr>socket=tcp</a>:<a href="http://192.168.152.30:5060" target="_blank">192.168.152.30:5060</a><wbr>></div>
<div>Record-Route:
<<a class="m_7988023869890242840moz-txt-link-freetext">sip:192.168.152.30;transport=<wbr>tcp;r2=on;lr=on;ftag=<wbr>6210295535;ngcplb=yes;socket=<wbr>tcp</a>:<a href="http://192.168.152.30:5060" target="_blank">192.168.152.30:5060</a>></div>
<div>FROM: "Chris Hoffmann"<<a href="mailto:sip%3A%2B6490000431@test.com" target="_blank">sip:+6490000431@<wbr>test.com</a>;user=phone>;epid=<wbr>31D1C3F091;tag=6210295535</div>
<div>TO: <<a href="mailto:sip%3A%2B6421000004@192.168.152.30" target="_blank">sip:+6421000004@192.168.152.<wbr>30</a>;user=phone>;tag=3B920B2B-<wbr>5AB35EC4000D82E5-2232B700</div>
<div>CSEQ: 52 INVITE</div>
<div>CALL-ID: 80348062-6b75-444f-a5b0-<wbr>37ae28e958c1</div>
<div>VIA: SIP/2.0/TCP
192.168.152.18:50382;rport=<wbr>50382;branch=z9hG4bK601318c1</div>
<div>Allow: ACK, INVITE, BYE, CANCEL, REGISTER, OPTIONS,
SUBSCRIBE, NOTIFY</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 231</div>
<div>Contact:
<a class="m_7988023869890242840moz-txt-link-rfc2396E"><sip:ngcp-lb@192.168.152.30:<wbr>5060;ngcpct=<wbr>7369703a3132372e302e302e313a35<wbr>3038303b707278726f7574653d31></a></div>
<div><br>
</div>
<div>v=0</div>
<div>o=dcom 1521704645 1521704648 IN IP4 192.168.152.30</div>
<div>s=SIP Call</div>
<div>c=IN IP4 192.168.152.30</div>
<div>t=0 0</div>
<div>m=audio 30300 RTP/AVP 8 101</div>
<div>a=rtpmap:8 PCMA/8000</div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div>a=direction:both</div>
<div>a=sendrecv</div>
<div>a=rtcp:30301</div>
<div><br>
</div>
<div>------------------------------<wbr>-------------------</div>
<div><br>
</div>
<div>SIP 200 that works</div>
<div>------------------------------<wbr>-------------------</div>
<div><br>
</div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/TCP
192.168.152.18:50391;branch=<wbr>z9hG4bK82fc4379;received=192.<wbr>168.152.18</div>
<div>From: "Chris Hoffmann"<<a href="mailto:sip%3A%2B6490000431@test.com" target="_blank">sip:+6490000431@<wbr>test.com</a>;user=phone>;epid=<wbr>31D1C3F091;tag=9279e5c630</div>
<div>To: <<a href="mailto:sip%3A%2B64800000000@192.168.152.6" target="_blank">sip:+64800000000@192.168.152.<wbr>6</a>;user=phone>;tag=as50ee3235</div>
<div>Call-ID: 6dd401d6-8bbc-4fcb-9f37-<wbr>303d64293725</div>
<div>CSeq: 55 INVITE</div>
<div>Server: Asterisk PBX 1.6.2.11</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO</div>
<div>Supported: replaces, timer</div>
<div>Contact: <<a href="mailto:sip%3A%2B64800000000@192.168.152.6" target="_blank">sip:+64800000000@192.168.152.<wbr>6</a>;transport=TCP></div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 261</div>
<div><br>
</div>
<div>v=0</div>
<div>o=root 830409115 830409116 IN IP4 192.168.152.6</div>
<div>s=Asterisk PBX 1.6.2.11</div>
<div>c=IN IP4 192.168.152.6</div>
<div>t=0 0</div>
<div>m=audio 10070 RTP/AVP 0 8 101</div>
<div>a=rtpmap:0 PCMU/8000</div>
<div>a=rtpmap:8 PCMA/8000</div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div>a=fmtp:101 0-16</div>
<div>a=ptime:20</div>
<div>a=sendrecv</div>
</div>
<div><br>
</div>
<div>------------------------------<wbr>---------------------</div>
<div><br>
</div>
<div>Thanks,</div>
</div>
</blockquote>
<br>
</div>
</blockquote></div></div>