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  • 3rd party SIP and Connect

    We are trying to integrate a Connect system (Build 1801) with a 3rd party IVR server. The server acts as an SIP endpoint that registers to the Shoretel SIP proxy as an extension. The server is using Dialogic HMP SIP stack.

    Registration happens fine, SIP signaling and call setup is good, but we're getting no audio passed to the PRI (on an ST100DA which is also the SIP proxy). A wireshark capture from the ST100 reveals that audio is being sent just fine, but to port 1028 for some reason, and the ST100DA is just sending back ICMP port unreachable response.

    Further examination reveals that on the INVITE from the Shoretel, the SDP requests the RTP be sent to port 1028 for some reason.

    - Media Description, name and address (m): audio 1028 RTP/AVP 0 102

    Anyone know what would cause this?

  • #2
    This sounds like a bug. There are settings for SIP port range, but in my experience ShoreTel has followed them as set (and 1208 is definitely not in the default range). I would either open a TAC case, or bump your build to see if the issue is resolved.


    • #3
      Interesting! I had the same exact issue happen when I tried to help a customer integrate a "2N Helios IP Force" SIP door device with camera. While I would have loved to spend more time debugging, the customer had mounted them on the wall and made troubleshooting quite difficult. We were able to get around the issue by creating SIP trunks instead.

      My notes on my old ticket show the 400 series phone from ShoreTel trying to send RTP stream on port 1028.

      I say all that to say, ditch the extension - go SIP trunk.