Item #	Description	Possible Cause	Action Required
1	NMC server’s SIP contact header used old IP address that caused 32 seconds call dropped etc. Customer used cloud service for SIP trunk.	NMC IP address was not configured properly. Server is deployed in private network – not allow STUN server to exploit the external IP address.	Old IP address was manually set in /home/matrix/conf/sip_profiles/external.xml.extras – Need to replace the old IP with the new IP address to resolve the issue
2	After calling, I hear the ‘Welcome to …’ prompt, but USN does not accept any Access Codes.	License has expired Audio conference service may not be licensed. Eth0 is not in service so license does not recognize system.	Check Licensing Tab under System Configuration Page. Confirm no expiration date is shown. Then use Webmin custom command to check for a date in HMP portion of license. See section 5.3.7 search for string ‘ handler‘. See section 5.2 to confirm eth0 is in service
3	When I call into a conference room, I hear ‘echo’	If callers are very close to each other and using speaker phones, the audio from active speaker also get fed from a secondary caller’s phone into the bridge. This produces perception of echo.	Maintain a minimum distance of at least 5 feet between phone users, especially if speaker phones are being used.
4	When I call into a conference room, I hear ‘helicopter’ noise	There may be calls from a previous conference that are still connected between the PBX and the USN. This can lead to a circular loop that will build up the noise in a conference room.	Please make sure that PBX drops a caller’s line coming into the bridge after the caller has hung up. This condition can be verified by checking RealView. If there are calls that were not dropped, use the disconnect link on Realview to force a disconnect.
5	I tried to use iCAL button to send the meeting invitation, but no email was sent.	Your PC’s Outlook Calendar function may have been reset or set inappropriately. The iCAL facility needs to be enabled.	In Calendar, on the Tools menu, click Options, and then click Calendar Options. Under Advanced options, select the “When sending meeting requests over the Internet, use iCalendar format check box”.
6	Cannot dial out to bring additional people into a conference	The SIP trunk may be set up for inbound traffic only.	Please check with PBX/CO person. The trunks need to set for full duplex operation for this feature to work.
7	Hear bursts of noise after someone stops speaking in a conference.	HMP.Uconfig file is missing. HMP.Uconfig values need adjustment for that location	Reapply the license. This will reinstall the Hmp.Uconfig file Please bring this to the attention of XOP Customer Support for Assistance.
8	When I call into the bridge, it prompts me for an Access Code. The Access Code I enter gets rejected.	Access Code is probably for a scheduled conference so it is only valid during the time of the conference plus the Lobby time. The code will be rejected as invalid unless it is entered in the correct time window Some phones or phone systems distort DTMF, sending the system incorrect digits. The USN may not have correct license for the audio conferencing.	Confirm Access Code in question was tried during a time for which it was valid. Use the System Events in the GUI (BAD PIN) or tlog command to view the DTMF digits the system is detecting. Look for a single digit that was distorted into multiples or missing digits Please bring this to the attention of NTAC.
9	Intermittent choppy audio or other noise impairment in the audio conference. TIP: Request a recording of the conference when the noise is present.	Ethernet interface may be running in half duplex mode. Network not optimized for voice External Source injecting noise into the conference.	Confirm that all network elements the voice traffic crosses are configured to be full duplex. Confirm that the RTP packets coming from the USN have been marked as voice. Try muting the conference using 43#, does interference stop? If so use RealView to unmute lines one at a time to find the source of the spurious noise.
10	I set up my conference for recording but nothing is getting recorded.	Recording capability needs to be enabled on the admin side	Confirm that system has recording enabled for the moderator.
11	I am not able to schedule conferences	Scheduling capability needs to be enabled on the admin side	Make sure that ‘schedule port usage field’ is selected on System Configuration page..

Item #	Description	Possible Cause	Action Required
1	When I trigger a Firebar, the calls are being dialed out. If a call is picked up by an answering machine, I hear the answering machine welcome message	The NMC allows a moderator to select ‘disconnect answering machines’ on the advanced page for FB application	Select ‘disconnect answering machines’ settings. The system will automatically prevent answering machines from joining an ongoing audio conferencing. Alternately, select ‘any digit’ and only people who press a digit will be able to enter the conference.
2	I understand the system can SMS text messages	The system supports SMPP interface to external SMSCs.	This capability requires an account with a cellular network SMSC service provider.

Item #	Description	Possible Cause	Action Required
1	We are testing screen sharing. I can share my screen, but the other person only sees the below screenshot. I lowered the resolution as low as I could and still nothing. We are all internal here so firewall shouldn’t be an issue.	SSL Certification is not complete.	This is because of the certificate that did not include the bundle certification. For XOP WebRTC to work properly, it required bundle certificate. • Move current certificate to the BKUP folder • Append the bundle certificate to server.crt ‘cat sf_bundle.crt >> server.crt’ • Verify the file appended correctly. • Change wss and dtls-srtp.pem files in /home/matrix/certs/ • Back up these two pem file to BKUP folder • Remove wss.pem file • Create new dtls-srtp.pem with /home/wwwrun/etc/server.crt /home/wwwrun/etc/server.key > dtls-srtp.pem • Verify dtls-srtp.pem to have server.crt and server.key appended to it. • Make a symbolic link to the wss.pem ‘ln –s dtls-srtp.pem wss.pem’
2	Unable to upload a document	PDF is password protected and / or encrypted Document is too large	Upload of secure PDF’s is not supported. Maximum upload size is 10M or 300 pages
3	Document is ‘Distorted’ during upload / conversion. Fonts are changed, spacing is different, diagrams or objects are missing	All uploaded documents are converted to jpeg format to be displayed. Certain fonts and objects do not convert properly due to Microsoft Patent / Trademark issues	The best resolution is to use the Save as PDF feature that is standard in Office 2007 and 2010. PDF documents convert to jpeg without distortion. A third party pdf converter such as cute PDF can also be used.
4	Power Point animations / transitions are not working after upload.	Animations and Transitions are removed during the conversion process.	To preserve transitions and animations, the presenter must share their desktop and use slide show mode.
5	Remote Desktop View is not full screen for participant	Resolution difference between displays Aspect Ratio difference between displays	The participant can see the display in actual size. If the shared desktop is lower resolution or smaller, it will not expand past actual size The aspect ratio of the shared desktop is maintained so if a 4:3 display is shared to a 16:9 display or vice versa, the height will be a limiting factor that can prevent the image from being displayed full size.
6	Noticeable Lag on Participant side when desktop sharing	Insufficient Bandwidth at USN Insufficient Bandwidth at Participant PC	each participant. Total bandwidth required = 300Kbits/s * n where n is the number of participants. Run a speed test to determine available bandwidth The participant must have at least 300Kbits/s available. Run a speed test to determine available bandwidth
7	Methods for reducing the Bandwidth required for Web Conference.	Documents whenever possible. Use Region Share Reduce the resolution Reduce the color depth Simplify the Image	Uploaded documents require almost no bandwidth after the initial upload. The bigger the area shared, the more bandwidth required. If viewing the whole desktop is not required use Region Share The lower the resolution of the presenters screen, the smaller the bandwidth required. 1024 X 768 is a common size that should display well for all attendees Reducing the screen color depth to 16 bit from 32 bit halves the required bandwidth Switch the desktop to a solid color, instead of a complex image.

Item #	Description	Possible Cause	Action Required
1	Error appears on Monitor saying “Cannot Display Image” or similar	This can happen if the system was built connected to a monitor that has a resolution set to too high or too low for the current monitor. It can also happen the system was built with no monitor attached	Under /etc/X11 you will find a file xorg.conf. Delete this file and reboot the system with the monitor attached.

Item #	Description	Possible Cause	Action Required
1	Users obtained NMC license from LMS server, but NMC server rejected the license.	License was generated wrong Hardware ID.	License file is needed to generate for specific hardware address (case sensitive) so you need to Generate/download new licence file. Login as admin From NMC web portal, click Administration menu >> Licensing page (http://192.168.0.207/system/configure_license) Upload newly generated license

NMC is a sophisticated Internet appliance that supports Audio conferencing, Web conferencing and Emergency conferencing (Firebar) applications and other Value Added services. The NMC uses CentOS 7.0 Linux Operating System. Different servers are used based on the required capacity of the system. The server’s CPU provides all the Digital Signal Processing (DSP) functions required to combine the various audio streams into an audio conference. Concurrently the server’s CPU also receives and transmits screen scrapes and other functions required for supporting web conferencing. It interfaces with an IP PBX or VoIP Carrier via a VoIP/SIP trunk. The following are answers to some of the most common questions or problems users come across. Read common questions about: