DIVATEL BUSINESS THOUGHTS

written by Admin Maintenance at 2015-02-11

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General concepts on what our business is all about

What is Unified Communications System?

Unified Communications is the integration of all of your office communications, including phone, fax, chat and web mash ups. It is not necessarily a single product but it combines the power of many products into one, providing you with more features than proprietary systems that have fewer features.

What are the benefits of Unified Communications?

Unified Communications supports the competency of businesses in several ways. These include:
1. Better relationships with clients, suppliers and other business partners. Unified communications increase the profitability that employees receive messages on time.
2. Enhance teamwork among staff at headquarters and remote offices.
3. Improves employee productivity: Reduces the time spent by workers trying to contact other workers in the course of their duties.
4. Better business processes and reducing ‘human latency’: The integration of communications with business processes and enterprise applications enables a trigger from the applications to initiate communications automatically.

Can I integrate Asterisk with an existing PBX system?

Yes. Asterisk can be integrated to any existing PBX system with the use of VoIP gateways. These gateways connect traditional phone lines to your IP PBX. You can also connect your PSTN lines to the IP PBX using analogue telephony interface cards. These cards connect legacy phone lines, phones, and phone systems to Asterisk-run computers.

Is it possible to avail of advanced features for use with our current phone system?

If you are still using a legacy PBX, call features like call waiting, call forwarding and call transferring are possible but these come with extra costs. By converting your legacy PBX to an IP PBX or VoIP, you will be able to add more features to your phone system without the additional licence costs.

What are Digium Telephony interface cards?

Telephony interface cards are PCI or PCI Express expansion cards connect regular phone lines, analogue phones, and legacy phone systems to Asterisk.

There are different types of analogue cards. How they are configured depends on the number of analogue connections that you need. Digital cards, on the other hand, connect Asterisk-based systems to T1/E1/J1 and ISDN-BRI interfaces. Digital cards interconnect traditional telephony systems with VoIP technologies and can manage to add more digital call channels to your Asterisk system.

Should I choose an Open Source or Proprietary PBX?

Open Source PBX systems use software made available for free (like Asterisk). It is maintained by a community of developers and integrated to any existing PBX system. Open Source software is flexible and ideal for businesses who want to customise their software tailored to their business needs. In terms of hardware, the choice of brand and type depends on the client. Since open source software is developed publicly and in a collaborative manner, there are plenty of online helps available. Open source is reputed to be cheaper since there are no license fees to pay.

On the other hand, Proprietary PBX systems use proprietary software that is owned by a single developer. Proprietary software can be modified by the manufacturer at an additional cost. Most Proprietary software functions when used with hardware that is made by the same manufacturer. It also entails additional costs for every call feature you use as well as license fees and upgrade fees. The benefit of a proprietary PBX is that the code is consistent and the software comes with a comprehensive Service Level Agreement (SLA).

The benefits that you receive in using an Open Source PBX far outweigh the use of a Proprietary PBX, in terms of cost, flexibility and scalability.

Should I choose a Hosted or an On-Site PBX?

A hosted PBX is a phone system that resides in a data centre and is accessed via the Internet. A hosted PBX has lower cost of ownership as the provider houses the hardware and handles the technological aspects of the PBX. A hosted PBX is ideal if there are home workers or multiple offices or when staff moves around all the time. A hosted PBX is also suitable for a small business wishing to minimise capital expenditure.

An On-site PBX is installed on the client’s premises. This entails higher installation and maintenance costs. An on-site PBX is more flexible because it customises according to the needs of the business. An On-site PBX is also ideal for call centres because it has advanced call centre features. It is easy to integrate with CRM and other solutions residing on site.

What are IP PBX appliances?

IP PBX appliances are do-it-yourself phone systems that give small and medium-sized businesses a complete and efficient PBX system. There are different types of appliances to choose from depending on your business’ call volume.

There are also Asterisk-based IP PBX appliances which provide cost-effective solutions for businesses.

What is a VoIP Gateway?

A VoIP Gateway is the bridge between a traditional PBX to the IP Network. It converts PSTN lines to VoIP. There are analogue, digital and GSM VoIP gateways.

What are the differences of analogue, digital, and GSM VoIP Gateways?

Analogue VoIP gateways connect traditional phone lines to your IP PBX. Digital VoIP gateways connect one or more digital lines to each other (one or more E1/T1/J1/ISDN-BRI lines).

GSM VoIP gateways enable connections between IP, digital, analogue and GSM networks. This helps reduce the cost of IP to GSM calls.

What are VoIP phones?

VoIP phones are phones that use VoIP (Voice over Internet Protocol) which allow calls over the internet. VoIP phones are less expensive than traditional phones since there are several features in VoIP phones that are either available with additional cost or are absent in traditional phones. Long distance calls are also cheaper sometimes even free. VoIP phones include desktop VoIP phones, USB phones, wireless phones, soft phones, video phones and IP conference phones.

What is a VoIP termination service?

VoIP termination is routing a VoIP call to regular landlines or PSTN. It is when the call leaves the internet and converts the call into analogue signals since information transfer in VoIP are in the form of data packets.

What are the requirements for running Asterisk?

There are no strict requirements for running Asterisk but you need to have a standard x86 based processor. Although Linux is the officially supported OS of Asterisk, it can also run on other operating systems such as Windows and Mac. Internet connection is a requirement for it to run.

What is GUI?

GUI or Graphical User Interface is an interface that allows users to interact with computers (human-computer interaction). It makes use of icons, visual indicators and menus instead of text-based interfaces so it becomes easier for the user to navigate a certain application. Asterisk and its distributions use this kind of interface to make the customisation of a software application easier.

What is third party software?

Third party software is a software component developed, distributed and sold by a company other than the original developer or distributor of a development platform. Asterisk third party software are add-ons for Asterisk that are available from Digium and through other vendors. This software extends the power of Asterisk by providing additional functions.

What is failover? Do I need to have a failover solution?

Failover is when a computer switches to a backup, usually a secondary computer server, system, database or network when there is failure of the primary active server, system, database or network.

Yes. You need a failover solution for your business to run smoothly whenever a system or server fails.

What are SIP Servers?

A SIP Server is the main component of an IP PBX. Also called SIP Proxy or SIP Registrar, SIP Servers perform call set-up functions like call routing, authorisation and network access.

What is NAT?

NAT or Network Address Translation is the translation of a number of private IP addresses to a single public IP address, via a router/firewall, allowing to have a single public IP address for a number of devices on your LAN when it sends data through the Internet. NAT helps resolve the IP address shortage as well as increase the security of a private network.

Can you make free calls using Voice over IP (VoIP)?

Yes if you have an address resolution. Address resolution is when you know the IP address of the phone you want to call. But there are other things to consider like firewalls and NAT. Otherwise, calls over VoIP are not free but are actually cheaper than calls using PSTN. If you have a computer, Internet connection, microphones and speakers/headsets, you can use VoIP communication for free. An example of this is calling from Skype to Skype.

What are the factors affecting VoIP call quality?

1. RTP or Real-Time Transport Protocol
RTP is used in communication involving real-time data. It is used to stream audio and video over IP networks. Although RTP is supposed to send data in actual time, lapses in the transmission are inevitable since data is usually buffered. A buffer is a memory storage which temporarily stores data right before it is used.

2. Jitter
Jitter is the inconsistency in the timing of arrival of data packets. Jitter happens because of network sharing, LAN congestion, or improper queuing.

3. Latency
Latency or lag is the time it takes for a sent voice data to reach its destination. This is caused by slow network links leading to delays or echoes.

4. Internet Connection
Since VoIP uses the Internet to send data, poor Internet connection is a factor that affects VoIP calls. A broadband connection is needed in order to use VoIP since a dial-up connection is too slow for VoIP to work.

5. Equipment
A router may not work properly when using it for voice and data unless the router has the ability to prioritise VoIP traffic. Since call quality is affected if there are other users in the network, installing a VoIP-enabled router ensures better voice quality.