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<div class="moz-cite-prefix">Hi Chris,<br>
you should try editing the files:<br>
/etc/ngcp-config/templates/etc/kamailio/lb/kamailio.cfg.customtt.tt2<br>
/etc/ngcp-config/templates/etc/kamailio/proxy/kamailio.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="il">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"
cite="mid:CAN8EJkwYT0N_r7ZWaaejjVgMpX5MuFm-8-eLB-pbFkwtP4SEng@mail.gmail.com">
<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>---------------------------------------------</div>
<div>SIP/2.0 200 OK</div>
<div>Record-Route:
<a class="moz-txt-link-rfc2396E" href="sip:127.0.0.1:5062;lr=on;ftag=6210295535;did=1c9.d452;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=anl6aVVKfUJ5RXlpRmYJGDtnIxIzMyx2HhVwEzN6DR4kF0UZBlsNMQ--"><sip:127.0.0.1:5062;lr=on;ftag=6210295535;did=1c9.d452;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=anl6aVVKfUJ5RXlpRmYJGDtnIxIzMyx2HhVwEzN6DR4kF0UZBlsNMQ--></a></div>
<div>Record-Route:
<<a class="moz-txt-link-freetext" href="sip:127.0.0.1;r2=on;lr=on;ftag=6210295535;ngcplb=yes;socket=tcp">sip:127.0.0.1;r2=on;lr=on;ftag=6210295535;ngcplb=yes;socket=tcp</a>:<a
href="http://192.168.152.30:5060" moz-do-not-send="true">192.168.152.30:5060</a>></div>
<div>Record-Route:
<<a class="moz-txt-link-freetext" href="sip:192.168.152.30;transport=tcp;r2=on;lr=on;ftag=6210295535;ngcplb=yes;socket=tcp">sip:192.168.152.30;transport=tcp;r2=on;lr=on;ftag=6210295535;ngcplb=yes;socket=tcp</a>:<a
href="http://192.168.152.30:5060" moz-do-not-send="true">192.168.152.30:5060</a>></div>
<div>FROM: "Chris Hoffmann"<<a
href="mailto:sip%3A%2B6490000431@test.com"
moz-do-not-send="true">sip:+6490000431@test.com</a>;user=phone>;epid=31D1C3F091;tag=6210295535</div>
<div>TO: <<a
href="mailto:sip%3A%2B6421000004@192.168.152.30"
moz-do-not-send="true">sip:+6421000004@192.168.152.30</a>;user=phone>;tag=3B920B2B-5AB35EC4000D82E5-2232B700</div>
<div>CSEQ: 52 INVITE</div>
<div>CALL-ID: 80348062-6b75-444f-a5b0-37ae28e958c1</div>
<div>VIA: SIP/2.0/TCP
192.168.152.18:50382;rport=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="moz-txt-link-rfc2396E" href="sip:ngcp-lb@192.168.152.30:5060;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31"><sip:ngcp-lb@192.168.152.30:5060;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31></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>-------------------------------------------------</div>
<div><br>
</div>
<div>SIP 200 that works</div>
<div>-------------------------------------------------</div>
<div><br>
</div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/TCP
192.168.152.18:50391;branch=z9hG4bK82fc4379;received=192.168.152.18</div>
<div>From: "Chris Hoffmann"<<a
href="mailto:sip%3A%2B6490000431@test.com"
moz-do-not-send="true">sip:+6490000431@test.com</a>;user=phone>;epid=31D1C3F091;tag=9279e5c630</div>
<div>To: <<a
href="mailto:sip%3A%2B64800000000@192.168.152.6"
moz-do-not-send="true">sip:+64800000000@192.168.152.6</a>;user=phone>;tag=as50ee3235</div>
<div>Call-ID: 6dd401d6-8bbc-4fcb-9f37-303d64293725</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"
moz-do-not-send="true">sip:+64800000000@192.168.152.6</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>---------------------------------------------------</div>
<div><br>
</div>
<div>Thanks,</div>
</div>
</blockquote>
<br>
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