Incoming VoIP Features

From AAISP Support Site


Related Pages on the A&A Website:

Trunk and PBX Features

Our VoIP system can simple 'trunk' calls to/from your own PBX - it is perfectly normal for you to have your own office PBX, such as a FireBrick or an Asterisk server for example and for your equipment to provide you will all the 'PBX' type functionally that you require. However, you can also have SIP phones register directly against our servers and use the PBX functionality that we provide. Or, indeed, have a mix of the 2 - eg we can route calls to you by SIP (trunk), but we can also record calls.


Incoming Features Tab Overview:

These are the features on the Control Page, under the 'Incoming' Tab.

Recording

  • AAISP can record all call, just incoming or just outgoing, as either wav, mp3 or ogg.
  • Recordings are in stereo - with the 2 parties on separate channels.
  • Callers can be warned, with a pre-recorded announcement, that the call will be recorded

Queuing

  • Calls to this number will get ringing even if the target is busy
  • When the target is available, the call will be put through
  • Applies to SIP phones only

ACR

  • Anonymous Call Reject - callers who withheld their number will be rejected

Voicemail

  • Calls after a certain time can be sent to voicemail
  • Record your greeting by calling 1571 from the SIP phone registered to the number

Timezone

  • Callers will get a pre-recorded message saying that the person they are calling is in a different time zone, and will say what the time is there.
  • Set to Local to disable this feature

Syslog

  • You can set a syslog host to get syslog messages for all incoming calls to the number
  • Works for group rings and so on, even if you are busy or DND, etc.
  • It advises your number, the caller number, and caller name if known
  • Also see status page post

Profile

  • The time range that a number is in service
  • a number can have multiple time

Multiple Targets

Incoming call routing is configured on the Control Pages. Call Routing is based on setting the 'Target' - a number can have multiple targets, eg, can be routed to a SIP phone as well as a mobile, and multiple other numbers. Indeed, you can also register multiple SIP phones against the same number, and calls will go to all the SIP phones that are registered.

Delays

Each target can be given a delay (0-30 seconds). Eg, you can have your number ring your SIP phone immediately, and then also ring your mobile after 10 seconds.

Number Announce

Put a number in to here, and a message will be given to callers that the number has changed and the number entered will be read out as the new number. The call will then end.

Fail

Number to call if the call fails to get though to the configured endpoints (ie a registered phone). Fail number doesn't get used if you have voicemail set for a number, as the call will go to voicemail.

Transferring Calls

Transferring calls is supported, and with VoIP this is usually handled by your telephone. ie, your phone would have a Hold or Transfer button, which enables you to either Blind transfer or perform an assisted transfer.

Targets in Detail

A phone number can be given many targets, each target has a delay, so you can control which phones ring first etc.

SIP Phone

  • You can register multiple sip phones to our server. (single phones if you're on the legacy A server).
  • The Tag is 4 characters that will be prefixed to the number and shown on your SIP phone display (if it has one)

Mobile

If you have a SIM with us, then the call will also be sent to the SIM

Your Server

We can route calls to your own SIP server, fill in the details of your server here.

Also Ring

These are up to 10 other numbers that we'll send the call to. They can be other numbers you have with us, or can be any other number. eg mobiles, international etc - any number which you can normally dial from your account. The charge for the call will be the same as if you were dialling the call normally from your account.

Outgoing Tab Features Overview

Record

Same as Incoming feature

Centrex

This allows you to use the last 1, 2 or 3 digits of your phone number to call other numbers in your block. eg. if you have 2 numbers 01344400001 and 01344400002, then you can call eachother by using 001 and 002 if you set Centrex to 3. This is also used when you transfer calls between your numbers.

Presentation

This is the outgoing Presentation Digits that are set when a call is made from this phone number. eg, we can set something other than their phone number here. There is a charge and a process for this and we'll need paperwork to prove the number is yours. Contact Sales for more details.


Local Prefix

This is an area code, (eg 01344, 020) that will get prefixed to local numbers. Useful for people with an 03 number wanting to dial local calls without the area code

IP Lockdown

Registration and calls will only be allowed from this single IP address.


Access

Define the level of outgoing calls allowed. This is defined by setting the maximum call price per minute, set separately for National and International calls.

Bill Warning

We will send you an email when your monthly bill reaches this amount. This does not block the account, it's just an advisory message. We will send the email to the email set on the Number and also the email the Login.

General VOIP Security

VoIP accounts can be compromised , so care is needed to this does not happen. Please see our VoIP Security page for more information. VoIP_Security



Other Control Page pages:

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