Calls going fast busy instead of routing out to PSTN RRS feed

  • Question

  • I've got cloud connector installed and configured per the TechNet documentation.  However, when I attempt to make a call to a PSTN number form a Skype client (all we care about is outbound calling, inbound calls go through an auto attendant), the call immediately goes fast busy and fails.

    I've checked to make sure my FXO gateway is IP'd correctly and set up to receive TCP SIP traffic on 5060 (which is how I configured cloud connector).  I also set up the FXO gateway to talk to the mediation server on port 5068, but I did this through proxy settinsg as the gateway I have is asking for a username/password for SIP registration.

    My gateway is a Grandstream HT503 since we only have one PSTN line, it was the lowest cost option.

    Any suggestions on where to look next?


    Wednesday, September 27, 2017 11:59 AM

All replies

  • Some additional information on the issue.  I'm not sure its an issue with the grandstream.  I think there's an issue with the configuration.  When I run packet captures on the edge server traffic, I don't see any traffic coming inbound from the internet when I try and place a call that should route to the PSTN

    I've confirmed that the inbound firewall rules are correct and the DNS is correct..  I can connect to my edge server using telnet <FQDN> 5061 and telnet <FQDN> 443

    I followed the TechNet articles for configuring the configuration and when I check my user PSTN settings, the HybridPSTNSiteName is set correctly to the name of my site, and the HybridPstnSiteFqdn is set to the FQDN of my edge server.

    Also checking CsTenantHybridConfiguration shows all the correct settings for my CCE installation.

    Any ideas what I'm missing?

    Thursday, September 28, 2017 1:22 AM
  • Deleted
    • Proposed as answer by Alice-Wang Thursday, September 28, 2017 7:23 AM
    Thursday, September 28, 2017 2:42 AM
  • This is a Skype for Business online (cloud pbx) installation. Not sure how sniffing the logs on the client will help since the call is initiated but isn’t routing from the cloud to my edge server on prem for routing to my pstn line
    Thursday, September 28, 2017 8:54 AM
  • Deleted
    Thursday, September 28, 2017 9:51 AM
  • THanks for the pointers...  I'm seeing ms-diagnostics: 1003; reason="USer does not exist" when I try to call ANY external phone number.  It looks like the system is trying to find the users inside the SIP domain and isn't forwarding calls out at all..  

    Thursday, September 28, 2017 2:37 PM
  • Some additional information on some troubleshooting I did...

    I unregistered and then uninstalled my CCE appliance, and rechecked my configuration.  I ended up making a couple of changes:

    - In my original configuration I had tried to create the CCE topology as flat - everything on a single subnet as I didn't have any VLANs configured on my router.

    - In this new configuration, I added a VLAN to the router specifically for the external facing portion of the edge server so the topology matches the documentation:  Edge server with one IP address for external connectivity and another IP on the internal interface.

    Bottom line is, the problem is still the same.  However, at the end of the "Install-CcAppliance" script, I noticed a warning this time that read:

    Warning: Network Configuration has not been set up for this tenant.

    Is there some additional steps to take to address this warning and might that be causing the Office 365 tenant to not be properly sending PSTN calls to my CCE?


    Friday, September 29, 2017 10:50 AM
  • Some additional information...  I noticed that when I look at the voiceroutingpolicy assigned to the user I'm testing with, it shows InternationalCallsDisallowed.  This is OK since we aren't doing international calling.   When I look at that particular Voce Routing Policy, it shows that the PSTN Site Index is "0".  And the PSTN Usage field is also blank.

    When I look at the list of PSTN Sites I have configured, it shows my CCE Site has an index of 1.

    Could this potentially be the issue?

    If so:

    1) How do I modify the voice routing policy?  Set-CsVoiceRoutingPolicy is not an available command in the SfB Online installation


    Sunday, October 1, 2017 12:55 PM
  • And here's a reportError block from using Snooper to examine the client logs:

    - <reportError  xmlns="http://schemas.microsoft.com/2006/09/sip/error-reporting">
        -  <error  toUri="sip:+1##########@SIPDOMAIN.org;user=phone"
               <diagHeader>1003;reason="User does not exist";domain="SIPDOMAIN.org";source="sipfed2a.online.lync.com";OriginalPresenceState="0";CurrentPresenceState="0";MeInsideUser="No";ConversationInitiatedBy="0";SourceNetwork="0";RemotePartyCanDoIM="No"</diagHeader>
        - </error>

    I replaced the 10 digit US phone number with ########## and my actual domain with SIPDOMAIN.org

    Sunday, October 1, 2017 1:16 PM