Hi,
I have just put a SBC between mu IPO and my Skype for Business server. Everything seems to be configured properly on the SfB and SBC side. I can make calls from IPO phones to SfB users. However, I cannot make calls from SfB to IPO. The calls leave SfB server, are going thru the SBC and I can see them hitting the IPO but nothing happens after that. It looks like the IPO doesnt know which phone to ring.
Here's a call trace from SfB to IPO
955288045mS SIP Call Rx: phone
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 10.XXX.XXX.253:5060;alias;branch=z9hG4bKac72517411 6
Max-Forwards: 69
From: "XXXXXXX" <sip:+53999;[email protected];user=phone>;tag= 1c1405126161;epid=902F1D7780
To: <sip:[email protected];user=phone>
Call-ID: [email protected].253
CSeq: 1 INVITE
Contact: <sip:10.XXX.XXX.253:5060;transport=tcp;ms-opaque=7b537d9addec44ee>
Supported: 100rel,sdp-anat
Allow: ACK,CANCEL,BYE,INVITE,PRACK,UPDATE
User-Agent: M800B/v.7.20A.202.112
Content-Type: application/sdp
Content-Length: 356
v=0
o=- 893849865 1097867657 IN IP4 10.XXX.XXX.253
s=session
c=IN IP4 10.XXX.XXX.253
b=CT:1000
t=0 0
m=audio 7190 RTP/AVP 97 0 8 101 13
c=IN IP4 10.XXX.XXX.253
a=rtcp:7191
a=label:Audio
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
Extension 3860 is configured with SIP Name 3860 - SIP Display Name User A and Contact 3860.
I have Incoming Call Route for SIP Trunk as Any Voice to Destination .
I'm not sure where to look anymore.
Thanks
I have just put a SBC between mu IPO and my Skype for Business server. Everything seems to be configured properly on the SfB and SBC side. I can make calls from IPO phones to SfB users. However, I cannot make calls from SfB to IPO. The calls leave SfB server, are going thru the SBC and I can see them hitting the IPO but nothing happens after that. It looks like the IPO doesnt know which phone to ring.
Here's a call trace from SfB to IPO
955288045mS SIP Call Rx: phone
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 10.XXX.XXX.253:5060;alias;branch=z9hG4bKac72517411 6
Max-Forwards: 69
From: "XXXXXXX" <sip:+53999;[email protected];user=phone>;tag= 1c1405126161;epid=902F1D7780
To: <sip:[email protected];user=phone>
Call-ID: [email protected].253
CSeq: 1 INVITE
Contact: <sip:10.XXX.XXX.253:5060;transport=tcp;ms-opaque=7b537d9addec44ee>
Supported: 100rel,sdp-anat
Allow: ACK,CANCEL,BYE,INVITE,PRACK,UPDATE
User-Agent: M800B/v.7.20A.202.112
Content-Type: application/sdp
Content-Length: 356
v=0
o=- 893849865 1097867657 IN IP4 10.XXX.XXX.253
s=session
c=IN IP4 10.XXX.XXX.253
b=CT:1000
t=0 0
m=audio 7190 RTP/AVP 97 0 8 101 13
c=IN IP4 10.XXX.XXX.253
a=rtcp:7191
a=label:Audio
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
Extension 3860 is configured with SIP Name 3860 - SIP Display Name User A and Contact 3860.
I have Incoming Call Route for SIP Trunk as Any Voice to Destination .
I'm not sure where to look anymore.
Thanks
