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Dialin acces number RRS feed

  • Question

  • I have Skype for Business 2015 onsite.

    I have 2 virtual machines. One has is running the Edge role and the other is running the standard role with Mediation role co-located on it. All the non Enterprise voice stuff works well.

    I just signed up for a SIP trunk and am adding it as a dial-in access number.

    Where should the SIP trunk terminate to? The Edge server or the Standard/mediation server?

    Thanks!

    Monday, September 26, 2016 4:31 PM

Answers

  • In that case, run the debug logger and watch incomingandoutgoingcall.  Look at sipstack and search in the messages tab of snooper for that number to see what it's doing.

    Please remember, if you see a post that helped you please click "Vote" on the left side of the response, and if it answered your question please click "Mark As Answer". SWC Unified Communications This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, SWC, their employees, or other MVPs.

    • Proposed as answer by jim-xu Monday, October 10, 2016 8:37 AM
    • Marked as answer by jim-xu Tuesday, October 11, 2016 7:44 AM
    Tuesday, September 27, 2016 7:09 PM
  • Hi Paul, 

    Agree to Anthony s suggestions and isolation , time to check the ocslogger and find out where this call is heading to , you can enable the logger for ibound and outbound routing. 


    Linus || Please mark posts as answers/helpful if it answers your question.

    • Proposed as answer by jim-xu Monday, October 10, 2016 8:37 AM
    • Marked as answer by jim-xu Tuesday, October 11, 2016 7:44 AM
    Wednesday, September 28, 2016 10:07 AM

All replies

  • Hi,

    Usually to the mediation server role.

    cheers

    Monday, September 26, 2016 5:02 PM
  • That is what I figured. I have the ports identified that it is listening to open to it on the firewall. However, when I call the phone number it just rings and rings and rings. No answer.
    Monday, September 26, 2016 5:03 PM
  • Did you check the event logs. I'm not quite sure but I believe there most be ports open in both directions (initiated).
    Monday, September 26, 2016 5:05 PM
  • Is the sip provider certified for Skype for Business?  Are they using TCP and G.711?  Also, how are you pointing it to your mediation server?  Are you using a NAT'd address, or have you given one of your interfaces a public IP address?

    Please remember, if you see a post that helped you please click "Vote" on the left side of the response, and if it answered your question please click "Mark As Answer". SWC Unified Communications This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, SWC, their employees, or other MVPs.

    Monday, September 26, 2016 5:10 PM
  • I did and there were no errors. The Mediation server has no outbound restrictions.
    Monday, September 26, 2016 5:10 PM
  • Yes, it is circuitid.com, they are on the certified list.

    I am using DNAT and redirecting the 2 ports, one for TLS and one for TCP.

    Monday, September 26, 2016 5:13 PM
  • Make sure NAT is working the same in both directions, ensure that circuitid will work with the NAT, meaning that regardless of what your SIP packets say to send all media to the signalling address (your sip packets will tell circuitid to send the packets to your internal IP, which they won't have access to).  Make sure that they're sending the phone number in full E.164 format, and make sure that you're using full E.164 format when setting the dialin access number.  

    I'd suggest starting with TCP communications with circuitid.com, not TLS, just to make troubleshooting easier.  You can use Wireshark or the debug logger to watch the SIP packets to see what's going on.

    Try assigning the number to a user instead and see if you can get the phone to ring that way to elimiate the variable of a conferencing setup issue.

    Right now, there are a million things that could be configured incorrectly to cause this, you just need to whittle it down.


    Please remember, if you see a post that helped you please click "Vote" on the left side of the response, and if it answered your question please click "Mark As Answer". SWC Unified Communications This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, SWC, their employees, or other MVPs.

    • Proposed as answer by jim-xu Tuesday, September 27, 2016 7:30 AM
    • Unproposed as answer by Paul L Fisher Tuesday, September 27, 2016 6:33 PM
    Monday, September 26, 2016 5:21 PM
  • OK, I ran wireshark. The data is getting in the firewall and to the frontend/mediation server.

    I believe the issue is that our SIP addresses are f.last@domain.com and our uri is 1112223333@domain.com and Circuitid.com is sending 1112223333@lync.domain.com. They are doing that because our server external FQDN is lync.domain.com.

    Can we strip off the lync. inside of Skype and if so, how?

    Thanks!

    Tuesday, September 27, 2016 6:37 PM
  • What is your lineURI exactly?  Does it start with tel: and is it in E.164?  No you wouldn't strip the domain off within Skype.

    Please remember, if you see a post that helped you please click "Vote" on the left side of the response, and if it answered your question please click "Mark As Answer". SWC Unified Communications This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, SWC, their employees, or other MVPs.

    Tuesday, September 27, 2016 6:49 PM
  • As seen in my user record in Skype:

    Line URI:

    TEL:+14804986488

    Tuesday, September 27, 2016 6:58 PM
  • Wireshark shows this:

    SIP from address: sip:+14804986488@domain.com;user=phone
        SIP from address User Part: +14804986488
        E.164 number (MSISDN): 14804986488
            Country Code: Americas (1)
        SIP from address Host Part: domain.com
        SIP From URI parameter: user=phone

    With domain being my actual domain

    Tuesday, September 27, 2016 6:59 PM
  • That looks decent.

    Can you assign the lineURI to a user instead, does that make it work?

    Do you see a sip response being sent back out via wireshark?


    Please remember, if you see a post that helped you please click "Vote" on the left side of the response, and if it answered your question please click "Mark As Answer". SWC Unified Communications This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, SWC, their employees, or other MVPs.

    Tuesday, September 27, 2016 7:03 PM
  • Also, was PSTN conferencing turned on later, or set up at the start when Skype was installed?  If you enabled PSTN conferencing later in the topology builder, you might need to rerun step 2 of the deployment wizard.

    Please remember, if you see a post that helped you please click "Vote" on the left side of the response, and if it answered your question please click "Mark As Answer". SWC Unified Communications This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, SWC, their employees, or other MVPs.

    Tuesday, September 27, 2016 7:04 PM
  • It was done later. I did rerun step 2.
    Tuesday, September 27, 2016 7:06 PM
  • It is assigned to a person, me. I gave up the dial in conferencing until I can figure this out.
    Tuesday, September 27, 2016 7:07 PM
  • In that case, run the debug logger and watch incomingandoutgoingcall.  Look at sipstack and search in the messages tab of snooper for that number to see what it's doing.

    Please remember, if you see a post that helped you please click "Vote" on the left side of the response, and if it answered your question please click "Mark As Answer". SWC Unified Communications This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, SWC, their employees, or other MVPs.

    • Proposed as answer by jim-xu Monday, October 10, 2016 8:37 AM
    • Marked as answer by jim-xu Tuesday, October 11, 2016 7:44 AM
    Tuesday, September 27, 2016 7:09 PM
  • Hi Paul, 

    Agree to Anthony s suggestions and isolation , time to check the ocslogger and find out where this call is heading to , you can enable the logger for ibound and outbound routing. 


    Linus || Please mark posts as answers/helpful if it answers your question.

    • Proposed as answer by jim-xu Monday, October 10, 2016 8:37 AM
    • Marked as answer by jim-xu Tuesday, October 11, 2016 7:44 AM
    Wednesday, September 28, 2016 10:07 AM