What is the TPS and CTPS?

The Telephone Preference Service (TPS) is a list of consumers (including Sole Traders and Partnerships, except in Scotland) who have registered their wish not to receive unsolicited direct marketing calls. The Compant Telephone Preference Service (CTPS) is an extension of this to businesses.

It is a legal requirement that companies do not make such calls to numbers registered on the TPS. Companies who are reported to the TPS for breach of the above regulation are included in a monthly report sent to the Information Commissioners Office (ICO) the body responsible for enforcement.

How does TPS and CTPS work with the VoiceHost platform?

The TPS service checks the TPS register each time a telephone number is dialled and can be enabled on any Hosted PBX seat or SIP Trunk.

If the telephone number appears on the register the caller is notified and depending on the selected option below, can at their discretion, choose to continue their call.

You can enable either TPS (residential database) or CTPS (Businesses Database) or BOTH!

TPS/CTPS Possible Outcomes

What are VoIP Routes?

The flexile nature of IP telecommunications lends itself to multiple routes via multiple carriers. This offers resilience and flexibility to service providers and end-users alike.
VoIP quality and cost are the cornerstones to all service providers and are for the most part interchangeable. Typically the quality of the VoIP routes is more important than the cost for general business communications. Good quality routes result in a good quality call, and eventually customer satisfaction. Call quality can be monitored and measure as a MOS (mean opinion score) and routing can be shifted dynamically, whereas on the other hand, poor quality routes often result in complaints regarding the call quality.

What is CLI and why is it useful

CLI stands for Calling Line Identification. This is a presentation of a telephone number, typically in E.164 format to the callee by the caller. CLI is useful not only to identify a callee to a caller but also in routing and statistics. Cloud telephony platforms such as VoiceHost use CLI to router calls and generate metrics.

VoIP Routes and CLI 

CLI Route: In CLI Route the recipient will receive the call after it goes through a chain of carriers. In this type of VoIP Route, the Caller ID is visible. This VoIP Route is commonly called the White Route. CLI Routes are considered to be authentic and called 'White Routes' within industry. White routes always aim to display the caller ID when the call connects to the recipient. The Caller ID visible to the receiver is actual information and not some random generic number. White routes are considered to be “legal” in terms of route termination on any end. All the connections associated with this route termination is in accordance with the law and also well connected to the telecom infrastructure of the concerned country. These legal routes containing the caller information are also of the best quality among all the other routes. The audio quality these routes offer is of international standard. And in addition to this, they are reliable in terms of connectivity and any other technical aspects. The cost of CLI Routes is usually higher than other routes. And the reason behind its high price is primarily the quality. Although several other factors may also affect its final price tag.

Non-CLI Route: The Caller ID is not visible at the call destination in this type of Non-CLI or Grey Route. A GSM gateway can terminate this type of calls. The NON- CLI Routes are also known as grey routes. The calls terminated through non-CLI routes do not display any information about the caller. It usually displays a blocked call or some random number. Many countries have restrictions on grey-routes. Due to the limited information available on the caller, the questions arise on its legality and its termination or any possible violations. Usually, these routes have impromptu set-ups. In addition to that, they further use GSM or cellular service gateways for traffic handling and call termination over the destination telephone network. On the aspect of call quality, the non-CLI routes also result in decent quality calls. Although usually, the quality of cli routes are better. But, as previously mentioned, there are many factors which affect the VoIP Call quality apart from the type of route.

Direct Routes: When the Route Provider has a piece of terminating equipment, it forms a direct route as there are no intermediaries. The call passes directly from the device to the provider before reaching its final destination. In reference to the legality of CLI and NON-CLI VoIP Routes, the CLI Routes carrying and displaying the caller information are legal. While the NON-CLI routes which do not reveal the caller information are termed as illegal.

Grey Black and White Routes: Incumbent networks like BT in the United Kingdom or any other government owned or ex-owned companies exercise monopoly in their respective telecom markets often keeping prices high unless the market is well regulated. To counter this high-cost issue, traffic is sent to country X via Internet (IP). The local PSTN terminates these at rates much lower than the international call termination charges. Therefore, on one end the routes are illegal or “black”, that is in other countries. But on the other end, they are “white”, which is in country X. And as a result, the route with one end black, one end white is then called a grey route.

Configuration Guide for Linksys ATA PAP2T

1. Plug the ATA into the network, plug into analogue phone, then plug in the power.
2. Give it a minute or two to start up, then dial **** on the analogue phone, then when the prompt plays, dial 110# to obtain the IP address of the ATA. (IP address will be in the format 192.168.x.x where x is between 0 and 255)
3. Plug the IP address into a web browser on the computer.
4. Click on the “Admin Login” on the right-hand side of the screen.
5. Click on “Line 1”

Change the following values:

• Line Enable: yes
• Sip Port: See Control Panel for Details
• Register Expires: See Control Panel for Details
• Proxy: See Control Panel for Details
• Display Name: <Your Name> 
• User ID: <Extension Number> See Control Panel for Details
• Password: <Extension Password> See Control Panel for Details
• Preferred Codec: G711a

Why use the fax to email feature?

Once you have understood the advantages of sending fax messages over and IP network or the Internet instead of the traditional phone line, you will want to know how you can achieve that. One important thing to note in IP Faxing is that fax messages are not sent to and from phone lines as traditionally, and hence do not require phone numbers, but are sent to and from IP addresses. Let’s see how you can fax over IP in the three ways.
Using a fax gateway, through the T.38 protocol. This is also called real-time faxing, i.e. the fax messages are sent and received at once, without having to wait and stored. It works, on the outside, like traditional faxing.

If you point your virtual fax number (any number can be pointed to the VoiceHost fax service) you will then be able to receive faxes to any emails specified in the control panel.

Increase Security

- Optional password protection of the PDF file that is emailed allows only the relevant person(s) to open and read the fax.

Setting up the feature

1. Navigate to 'Telephone Numbers' within the control panel (see below)
2. Select the destination of the number as 'Fax'
3. Enter the required email address and password (optional) as required.

How fast does my internet connection need to be?

Data connections have two components – bandwidth and data allowance.

• Bandwidth is measured in MegaBITS per second (Mbps) - 1024 Kilobits = 1 Megabit
• Download allowance is measured in MegaBYTES (MB) - 1024 Kilobytes = 1 Megabyte

How much data would be used based on 1-hour audio call using the G711 or G722 codecs with allowance for an IPv6 overhead?

• Each Phone, when in use, requires 80-100Kbps(down)/80-100Kbps(up) per voice call.
• Each phone, when in use, will use 38 MB of your data allowance per hour.

Example: An ADSL Connection (Asymmetric Digital Subscriber Line) may have connection speeds of 3.2Mbps/.8Mbps. Therefore, the available bandwidth at any one time is 3.2Mb for download, and .8Mb for upload.

Calculating Download Capacity of the above example:

Formula: (No. of Mbps * 10) = Possible number of concurrent calls over available download bandwidth.

3.2* 10 = 32

Calculating the Upload Capacity of the above example:

Formula: (No. of Mbps * 10 = Possible number of concurrent calls over available upload bandwidth.

.8 * 10 = 8

Therefore, because the most calls possible on the existing upload capacity is 8, only 8 phones should be used on this data connection.
If the 8 phones were in use for 10 hours each, the total amount of data allowance used would be:

8 * 10 * 38MB = 3040MB

To summarise, an ADSL Connection with speeds of 3.2Mbps/.8Mbps will support 8 phones, and use 3040MB of data over 80 hours of talk time with NO Available bandwidth left for anything else.

Configuration Guide for Aastra 6753i SIP Phones

Factory Reset (If pre-used)

• On the handset press the button printed below:
• Scroll down until you see “Admin Menu” and hit the right arrow key located under the display.
• The phone will ask for a “”Admin Password” as default this is “22222” or alternatively enter the admin password.
• Press the down arrow to enter the menu.
• Scroll down to option “4 Factory Default”, and press the right arrow twice, this will then carry out a factory reset and reboot.

Obtain IP address

• Press the “Spanner” key.
• Scroll right to “Phone Status”.
• Press the right arrow to enter into this menu.
• Press down once until “IP&MAC Addresses” is displayed and press the right arrow.

Take note of the IP Address

On a PC

• Enter the IP address into the internet browser address bar.
• Then log in as below (Username: admin Password:22222)
• Once Logged In, Click Global SIP located in the left-hand side menu.

Enter the details into the fields with the Green Dots

The username & password will be given & should look something like this:

Username: XXXXX*XXX

Password: (Password Is Case Sensitive)

Username :XXXXX*XXX

Password : XXXXXXXXXXX

The next section covers Basic SIP settings.

The details to enter in the fields marked with the green dots are as follows:

Proxy Server:                See Portal for Details

Proxy Port:                   See Portal for Details

Registrar Server:           See Portal for Details

Registrar Port:               See Portal for Details

The next section covers Advanced SIP settings.

The Following sections need altering

Local SIP UDP/TCP Port:                  See Portal for Details

Local SIP TLS Port:                            See Portal for Details

The final section is RTP Settings:

The RTP Port field needs altering to 3000

Scroll down to the bottom and click “save settings.”

No more settings need altering after this and you can close the browser down. The phone should auto-register.

Legacy CDR API - Sunset 2024 (deprecated)

URL and Access: Contact Support
Username: <username> has to be an enabled user within User Manager that has the Call Records privilege
Password: <password> is password of the <username>
Date Range: 
start=YYYY-MM-DD
end=YYYY-MM-DD (has to be after start date)

-<CDR>
<Status>0</Status>
<StatusMsg/>
-<CDREntry>
<Date>2017-03-03 22:00:42</Date>
<Source>07801123456</Source>
<Destination>Ext: 224</Destination>
<Duration>00:00:11</Duration>
<Direction>Inbound</Direction>
<InboundTelephoneNo>01234567890</InboundTelephoneNo>
<UniqueID>hpbx05-1488533999.45843</UniqueID>
</CDREntry>
</CDR>