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Answers to Frequently Asked Questions

Hello!

VCP

  1. Enter a unique extension that you will not be using on your PBX.
  2. Enter description (Sales Team, Technical Support ETC)
  3. Select Ring Group, sometimes referred to as Hunt Group:
    • Type of call: I.e. If you want all devices to ring at the same time (Simultaneous)
    • Sequential: to have Devices ring in a designated order).
  4. Choose Greeting. You will see a drop box showing you all the announcement choices.
  5. Voicemail Setting. Chose one of these four choices. Disable the call will disconnect the call, Email Address, Put into User Mailbox, Switch to Menu will transfer the call into an automated Attendant Menu.
  1. Under Domain Settings
  2. Go to Menu Tree
  3. Click on PBX Menu
  4. Create New Menu Description
  5. Click on the description e.g. Night Message
  6. Press the create new root button.
  7. Creating new action added the schedule
  8. Action sent messages to Users EXT.
  9. Prompt type change to “ Custom Recording”
  10. Save Menu
Note: If you have more than one Root Menu you must have a scheduled defined so that the system knows which menu to use.
  • Go to Domain Settings
  • Select User Accounts
  • Go to the blue person
  • Select the Applications Tab
  • Click on Attendant Console
  • Click Settings
  • Select the desired extensions and click on Display, Monitor and ring
  • Click Save
  • Shared line appearance (SLA) is used for small business customers with Key System handsets having multiple buttons representing individual lines. SLA allows employees to monitor the phones of the user on the account (similar to busy lamp field functionality) and also let them pick up the call or park those calls as desired. A user can answer calls on other extensions when their colleagues are asway-or put call son hold for another phone. Administrative assistants can use SLA to monitor employee calls and answer park or transfer them on an employee’s behalf.
Redirecting your calls to an alternate number such as an office phone or user’s cell phone or colleague’s number.
  • Dial *72 and then dial the 10-digit number to forward your calls.
  • To remove call forwarding dial *73

Device setup

  1. Accordion Sample Description1. Reboot device – disconnect power cord for 3 seconds and plug it back in. When you see the words “Starting Application”, press cancel
  2. When the Welcome screen appears, press the SETUP Softkey.
  3. If you wait too long, you’ll get an Updating Configuration message that will tie up the phone system for a few minutes
  4. Enter Default Password 456 – Press OKAY
  5. Select Provisioning Server
  6. Select DHCP Menu
  7. EDIT Boot Server
  8. Locate round navigation button and Press Right Arrow until STATIC is highlighted
  9. Press OKAY and EXIT
  10. EDIT Server type
  11. Press Right Arrow until HTTP is highlighted
  12. Press OKAY
  13. Highlight will then move to the next line: Server Address
  14. Press EDIT
  15. Press button located below the “ASKEY-KEY” (ASCII key/ a→ 1A)
  16. From the Keypad enter 38*102*250*15
  17. Press OK when done
  18. Press EDIT for the Server User
  19. Clear field by pressing DELETE button or press Left Arrow
  20. Leave blank and Press OKAY
  21. EDIT Server Password – Clear field by pressing DELETE or Left Arrow.
  22. Leave blank and Press OKAY
  23. Scroll down to “Tag SN to UA”
  24. Press EDIT – Right Arrow to “ENABLE”
  25. Press OKAY - EXIT - Exit again
  26. SELECT SAVE & REBOOT
  27. Once you’ve completed the steps, wait a couple minutes for the configuration files to download.
  28. When phone registers, lift the handset and make a test call.
  • Here are some issues that can affect audio quality:
  1. Internet speeds are not fast enough
  2. Firewall or Router policy and rules are too restrictive
  3. Intermittent Internet problems
  4. Large file uploads and downloads
  5. Defective or malfunctioning phone
  • One-way or no-audio, sometimes referred to as dead air, may be caused by restrictive or improperly configured firewall rules/Router policies. Ideally, all SIP or VoIP firewall/router features should be DISABLED. Confirm your Internet Service Provider is not blocking Internet traffic ports.
  • If you have a Network Administrator, ask them to allow Internet traffic to and from your devices unrestricted. If firewall rules and/or Router policies must be in place, please use the following.
  • Or use a range that has four ports, the maximum number of simultaneous calls made. For example, 10 simultaneous calls will use 40 ports.
  • An echo can be caused by electromagnetic Interference coming from a nearby desk fan or heater.
  • Also, improperly holding the handset or ill-fitted headset or the phone speaker, handset, or headset volume is set too high. This is because the microphone can pick up sound from the earpiece and rebroadcast the sound back down the phone line.
  •  speech can be caused from slow internet speed when there is not enough bandwidth to handle one or more calls, or when someone is uploading or downloading large files. YouTube videos and streaming audio also use a lot of bandwidth. The thing to remember is that one call will use at least 120 kilobits per second.
  • The up and download speed should be at least 1 meg. If you have up to 4 phones, then a minimum of 5 megs is best. Best practice, is to always have at least twice the bandwidth needed
  • Static may be caused by malfunctioning phones or head/handsets. Faulty electrical source wiring may also cause static.
  • Surge protectors, UPS, or APC battery backup systems should be used. Moving the power adapter from one power outlet to another may resolve the problem.
  • If the device is five years or older it may need to be replaced.
  • In order to make and receive calls, your IP Phone or Analog Telephone Adapter must communicate with our registration server. When a call is made, the IP Phone or ATA will send a signal to our server.
  • Problems with your Internet will cause problems with your phone.
  • Most IP Phones and ATAs will give a dial tone when the device connects to your home or office network. Just remember, hearing a dial tone does not mean you can make a call.
  • A busy signal is also an indication that the phone lost registration and is no longer communicating with our server. Or if someone calls you and they get your voicemail greeting instead of ringing.
  • Check the internet connection by opening a website. If you can’t connect then your device lost registration
  • Inquire with your Internet Service Provider, Office Network Administrator, or someone in the office who handles Internet and Network problems.
  • When your Internet is back up, unplug the power cord to your Phone or ATA for at least 10 minutes. After you plug it back in, wait a couple of minutes for the device to communicate with our servers.
  • Lift the handset up. If you get a dial tone, make a test call. If you hear ringing, all is well. If not, wait a few more minutes before trying again. If you still get an immediate busy signal after entering the last number, you may still have problems with your Internet.
  • The important thing to remember is that the phone or ATA will only work when the Internet is working.
  • If the Internet is working and you checked and rechecked all the wiring and connections to the phone or ATA, give us a call at 800-254-3109. We’ll be more than happy to help!
1. How to make a Supervised Transfer using a Polycom Phone
  • When on an active call, press the Transfer soft key. The call is then automatically placed on hold.
  • Dial the extension number or telephone number. Wait for the call to be answered. If the call is accepted, press the Transfer button again.
  • If the call is not accepted or you want to cancel the transfer attempt press CANCEL.
2. How to make an Unsupervised Transfer with a Polycom Phone
  • To make an unsupervised or cold transfer using a Polycom phone.
  • When on an active call press the Transfer soft key. The call is automatically placed on hold.
  • In the upper left corner of the display screen you will see the word BLIND. Select or press the softkey. Dial the extension number or telephone number. The system will dial the number and transfer the call.

Orders

  • All new orders should be sent to orders@xcastlabs.com
  • Once order is received, please allow 24 - 72 business hours for processing
  • All new DID and TFN requests should be sent to orders@xcastlabs.com
  • Once order is received, please allow 24 - 72 business hours for processing
  • All DID and TFN porting requests should be sent to lnp@xcastlabs.com
  • Standard porting Intervals for DID’s is 5 – 7 Business Days.
  • Standard porting Intervals for TFN’s is 3-5 Business Days.