[Spce-user] MS Lync with SipWise

[Chris Hoffmann]

Awesome, thanks! It working perfectly.

On 23/03/2018 12:33 AM, "Andrew Pogrebennyk" <apogrebennyk at sipwise.com>
wrote:

> Hi Chris,
> you should try editing the files:
> /etc/ngcp-config/templates/etc/kamailio/lb/kamailio.cfg.customtt.tt2
> /etc/ngcp-config/templates/etc/kamailio/proxy/kamailio.cfg.customtt.tt2
> (create them from tt2 according to handbook if they don't exit)
> and add the line
> modparam("rr", "enable_full_lr", 0)
> in both. This should do the trick!
>
> Regards,
> Andrew
>
> On 03/22/2018 08:51 AM, Chris Hoffmann wrote:
>
> Hi,
>
> 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.
>
>
> SIP 200 that doesn't get recognised by Lync
> ---------------------------------------------
> SIP/2.0 200 OK
> Record-Route: <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-->
> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=6210295535;ngcplb=yes;
> socket=tcp:192.168.152.30:5060>
> Record-Route: <sip:192.168.152.30;transport=tcp;r2=on;lr=on;ftag=
> 6210295535;ngcplb=yes;socket=tcp:192.168.152.30:5060>
> FROM: "Chris Hoffmann"<sip:+6490000431 at test.com;user=phone>;epid=
> 31D1C3F091;tag=6210295535
> TO: <sip:+6421000004 at 192.168.152.30;user=phone>;tag=3B920B2B-
> 5AB35EC4000D82E5-2232B700
> CSEQ: 52 INVITE
> CALL-ID: 80348062-6b75-444f-a5b0-37ae28e958c1
> VIA: SIP/2.0/TCP 192.168.152.18:50382;rport=50382;branch=z9hG4bK601318c1
> Allow: ACK, INVITE, BYE, CANCEL, REGISTER, OPTIONS, SUBSCRIBE, NOTIFY
> Content-Type: application/sdp
> Content-Length: 231
> Contact: <sip:ngcp-lb at 192.168.152.30:5060;ngcpct=
> 7369703a3132372e302e302e313a353038303b707278726f7574653d31>
>
> v=0
> o=dcom 1521704645 1521704648 IN IP4 192.168.152.30
> s=SIP Call
> c=IN IP4 192.168.152.30
> t=0 0
> m=audio 30300 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=direction:both
> a=sendrecv
> a=rtcp:30301
>
> -------------------------------------------------
>
> SIP 200 that works
> -------------------------------------------------
>
> SIP/2.0 200 OK
> Via: SIP/2.0/TCP 192.168.152.18:50391;branch=z9hG4bK82fc4379;received=192.
> 168.152.18
> From: "Chris Hoffmann"<sip:+6490000431 at test.com;user=phone>;epid=
> 31D1C3F091;tag=9279e5c630
> To: <sip:+64800000000 at 192.168.152.6;user=phone>;tag=as50ee3235
> Call-ID: 6dd401d6-8bbc-4fcb-9f37-303d64293725
> CSeq: 55 INVITE
> Server: Asterisk PBX 1.6.2.11
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> Contact: <sip:+64800000000 at 192.168.152.6;transport=TCP>
> Content-Type: application/sdp
> Content-Length: 261
>
> v=0
> o=root 830409115 830409116 IN IP4 192.168.152.6
> s=Asterisk PBX 1.6.2.11
> c=IN IP4 192.168.152.6
> t=0 0
> m=audio 10070 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> ---------------------------------------------------
>
> Thanks,
>
>
>
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