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Answers to Frequently Asked Questions

VCP

  1. Select the Call Groups feature in VCP, under Domain Settings.
  2. Enter the desired 3-or 4-digit extension in the Extension field. If the User Account extensions are all in a standard range like 101-105, we suggest using a higher range, e.g. 201, 301 etc.. to differentiate the Call Group from the User Accounts.
  3. Enter a description for your Call Group in the Description field, e.g. "Main Call Group", "Sales Team", or "Front of House" etc..
  4. Select the Service Plan. This feature is typically already set by default.
  5. If your PBX domain uses Divisions , you will see a drop-down box with the available Divisions below the Service Plan box. Select the Division from the drop down box.
  6. Select the Country / Area Code. The default is populated from the Main PBX area code , but it can be set to Custom using the radio button and Area Code field.
  7. Select the call group distribution type from the Type drop-down field. A Simultaneous ring type will ring all phones in the call group simultaneously, and a Sequential ring type will ring phones in order of precedent.
  8. Select the call group line ringing behavior from the Lines to call drop-down field. Dial idle lines only will not ring at lines that are busy or unavailable. Dial idle lines first will ring the available/not busy lines in order of precedent first, and then attempt the busy/unavailable lines to try and complete the call. Dial all lines will ring all lines in order of precedent, regardless of their availability.
  9. Select the call group ring tone from the Ring Style drop down. There are four options present, and each plays a different ringing tone depending on the IP phone device type. This is useful if your end-user needs to differentiate between direct calls to their User Account extension and calls from the Call Group based on the ring tone.
  10. Define the number of total rings from the Rings Per Phone fields. If you chose Simultaneous as the Type of call distribution, define the total number of rings that you would like for all phones to ring simultaneously before reaching Voice Mail.

    If you chose Sequential as the Type of call distribution, define the number of Rings Per Phone in the first field, and the total number of Rings Per Call in the second field. The total number of rings should equal the number of Rings Per Phone multiplied by the number of lines you would like the call to ring at, e.g. Rings Per Phone set to 3, and there are 5 extensions - 3 * 5 = 15 Rings Per Call.

  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. On your phone select the "Menu" key or soft-key
  2. Settings-->Advanced-->Admin Settings-->Network Configuration
  3. Set DHCP Menu-->set Boot Server to "Static" (you may need to push the right arrow key to scroll through the options)
  4. Set DHCP Menu-->set BootSrv Type to "String" (you may need to push the right arrow key to scroll through the options)
  5. Set Server Menu-->Set the Server Type to "HTTP" (you may need to push the right arrow key to scroll through the options)
  6. Set Server Menu--> Set the Server Address to "vernon.siptalk.com" (For "." is press the "*" key)
  7. Set Server Menu--> Set the Tag SN to UA to Enable
  8. Save Configuration - The phone should now reboot
  9. Please test to make sure the phone can make an outbound and inbound call with audio flowing in both directions.
    What can cause audio problems?
  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
    Here's our suggested solutions:

  • 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.

  • Choppy 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.