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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:3860@10.XXX.XXX.252;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;ext=53999@XXXXX.local;user=phone>;tag= 1c1405126161;epid=902F1D7780 To: <sip:3860@10.XXX.XXX.252;user=phone> Call-ID: 3952056142192018104148@10.XXX.XXX.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 |
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