What is SIP and SIP Trunking?

Many businesses are switching from the conventional, expensive, inflexible phone lines
(ISDN, PSTN) to a more modern and digital substitute: SIP trunking Although SIP trunking has been available for almost a decade it is yet a new approach to many. Therefore, there are a lot of unanswered questions for business owners about a transition to internet-based communication solutions. Here are the answered to some of those frequently asked questions.


What is SIP and SIP Trunking?
Session Initiation Protocol (SIP) is a prevailing and dominant transmission communication
sessions protocol. SIP trunking is a method for delivering voice and various unified communication services over the web that works when a SIP communication system and handset (PBX) is enabled. SIP trunks are a future replacement for standard phone carrier services, Pots, BRI, PRIs, services (ISDN, PSTN).

What is a SIP Channel?
SIP channel is a passage for receiving or sending a call. In other words it is a kind of standard phone line. Every SIP trunk has the ability to support any types of SIP channels.

Why Do Businesses opt for SIP Trunking?
SIP trunking gives businesses a chance to pay a low monthly fee per SIP channel and
benefit from an almost free calling system on a national or international scale. More than that, it also gives the customers the option to add ore remove channels as required and to completely control and manage their telecommunications needs as their businesses change. Additionally, SIP simplifies hosting unified communication options such as videos, instant electronic messaging, and application integration.

Is SIP a similar thing as VoIP?
Voice over Internet Protocol (VoIP) is a protocol used for voice decision web remodeling.
SIP is one of the various protocols used to achieve VoIP.

How Do SIP Calls Sound?
When a SIP provider executes Tier-1 networks with the right configuration on their router,
the quality of a SIP call is the exact same mimic of that of a regular phone line. In some cases SIP provides an even better call quality. As SIPs support various CODECs it can also
swiftly adapt the latest HD voice call qualities.

How difficult is it to Setup and Maintain? In order to setup and maintain a SIP solution there’s no need for a telecommunication
professional. All you have to do is to sign up with a SIP trunking provider that provides a
user-friendly administration portal, documents the process accurately and has a committed support team. To begin using SIP you will need a candid PBX and router configuration.

Are there Any Up-Front Costs?
It’s highly probable that SIP can be optimized to work with your current PBX. With SIP
processing enabaled, most PBX systems have been advocated to users in the last decade.
With a business-grade router at place there’s a high potential that it’s already has been integrated with the standard service requirements to provide the best current SIP solution for your business. On the other hand, the SIP solution provider can impose some initial setup fees for the supported channels. There could be some further charges for providing extra services such as incoming toll-free service, DIDs or etc.
So, indeed, there are some initial costs, however, compared to a conventional just out of the box standard communication system, these charges are yet insignificant. Hopefully we have been able to address some of your primary concerns. Please feel free to contact us for further familiarization with our services and request a free trail.

10 Benefits of Hosted PBX

With the constant advancements in the IT sector, as businesses try to adapt Cloud-hosted phone systems, it is a question many business owners or IT managers whether or not Cloud-Hosted PBX is the solution for your business or will it give its place to the next innovation that comes round. If you already have a phone system in place that is meeting your current requirements you might ask yourself this question” Does the cost worth the trouble?”

In this post, we will look at some of the many advantages of Hosted-PBX to help decision making easier for you.

Some of the advantages of a Hosted-PBX are as follow:

1)  Highly Cost-effective

Hosted-PBX is a kind of means that do not require the purchase of complicated hardware as the entire phone system is hosted and controlled in the Cloud. Therefore the major cost will only be the services you need without having to invest in capital costs and the technical aspects of setting up a system. From the Service Provider outlook, the costs are generally divided amongst the numerous users which on a bigger scale helps remarkably reduce the per unit usage/subscription fees for the customers in comparison to the cost of using a fixed line.

Hosted-PBX is not geographically limited that allows business such as Call Centers to conduct a substantial number of long-distance calls. This enables the entire company to form a single consolidated communication infrastructure regardless of their geographical location that consequently makes calls amongst various branches look like internal calls.   The combination of call plans that include unlimited calls to various mobile numbers will lead the average cost per call to significantly reduce.

2) Time-effectiveness

Generally speaking the overall installation of a typical PBX including ordering hardware, finding a professional to assess the call flow specifications and on-site installation can take up to a couple of months. On a Cloud Hosted-PBX though, installation is swift and analyzing call flow requirement can be done on a graphical user interface which can promptly be accessed via your desktop computer. You can add a single phone line or a couple of them quickly by clicking the configuration in the software. Also, the server maintenance takes less time and equipment can be upgraded faster as the service provider support team works 24/7 to ensure there are no downtimes.

3) Integrate your old/existing phone system

In the case that you want to maintain your existing solution but want to upgrade to Cloud-Hosted PBX in case of a branch in a new city or a new office, this can be easily achieved as Hosted-PBX allows for legacy system integration or device with a few simple clicks. Even better if your legacy system supports SIP trunks/VoIP there’s no need to purchase additional hardware.  Cloud-hosted-PBX gives you flexibility on how to route your calls with easy interface with existing infrastructures. Hosted-PBX configuration allows users on both systems to send and receive calls to and from each other while the legacy PBX users enjoy even cheaper calls via VoIP routes. This is especially an advantage in case of businesses that would like to maintain their existing PBX but also want to enjoy affordable prices via cheaper trunk lines.

4) Flexibility

Hosted-PBX is not limited geographically which allows you to simply use your VoIP phone, the smartphone with a VoIP application, or a desktop computer from any maker to connect from around the world. This gives the employees an option to access from anywhere on any device of their choice using the same office number assigned to them. You can transfer office calls from your office phone to your smartphone and answer calls in-transit. This feature is empowering because it increases your reachability to customers via a virtual connection or a toll-free number such as 03/0800 on the Hosted-PBX from any remote location or your office.

5) Advanced features

Even when you are using basic equipment you’re no longer restricted by simple features on your phone or in-house PBX. Hosted-PBX makes it possible to access all the available features by your service provider with no additional fees. These features include but are not limited to Call control, Call Recording, Queue, Voice Menu, Time Conditions, Wallboards and etc. The service provider gives you access to these features on a web portal where you can use the new features and upgrade the existing ones.

6) High-quality Voice calls

Over the years improvements on Quality of Service (QoS), faster internet speeds, and better compression rates have contributed to improving voice communication means via the internet. Before VoIP installation, the service provider can commence various tests to assess your internet connection speed and quality in order to offer the best VoIP solution that meets your requirements. Hosted-PBX utilizes voice prioritization and other packet loss resilience schemes to provide the highest quality calls for you. Nowadays service providers use their own networks such as Fiber Broadband 4G or ADSL in order to apply QoS schemes from the headquarter to customers and assuring the highest call quality.

7) Scalability

Whether your business is growing or you decide to cut on your business, you might want to add up or cut down your business requirements. Hosted-PBX helps you to pre-reserve resources for the possibility of future expansions; however, in fact, you are paying for unemployed resources which you may or may not use in the future expansions. With Cloud-Hosted-PBX you pay for your usage capacity as it enhances or as it diminishes. First, you can start with a small number of lines and then you can increase them as your business capacity grows. In other words, you can live up to the scale of your business as it meets your requirements. Cloud-Hosted-PBX allows you to scale up your business in high demand seasons by employing new lines and scale it down right after a festival or a major event by dismissing those lines.

8) Increased Redundancy and Disaster Recovery

One of the substantial concerns for the organization considering Hosted-PBX is reliability. While old PSTN can suffer from damage to device, hardware failure or power failure, and copper-wire damage, Hosted-PBX is relies on the internet and is managed by the service provider who has access to redundant resources including infrastructures and servers peering with various transit providers globally that can be used as backups to sustain business in case of natural disaster or manmade failure which is an added advantage over copper-wire landline.   

9) Real-time monitoring and Report Analysis

An advantage of Hosted-VoIP system is the accessibility of real-time statistics and motorization of all PBX activities including but not limited to ongoing and missed calls, call rates, active or available agents, and subscription costs through a web portal on a desktop or mobile. This enables business owners to analyze their employee’s performance, view instant call recordings and last but not least to make informed decisions about their business by using the reporting tool on PBX. The reporting tool helps to collect, collate, organize and present data using statistical tools such as average, and mean and graphical tools such as charts and graphs for better comprehension of the usage data.

10) Security

The security of Hosted-PBX is in general higher than in-house PBX. In VoIP systems, the service provider is in charge of assuring system security including software updates, firewalls and security patches. It’s the responsibility of the expert VoIP team to ensure system protection. Even in case of a security breach in your local network VoIP provider like VoIPLine telecom has defense algorithms in place to prevent large volumes of unauthorized calls.

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What is PBX and Why Cloud Hosted PBX is cost-effective?

PBX is a convenient tool for organizing day-to-day communications within and outside of your business. Have you ever had to press a certain number to speak with a certain person when calling your bank or your favorite food delivery to complain about the wrong order? If your answer is yes, then you have already interacted with the PBX.

Nowadays, communication is the key to success for many businesses. This could vary from communicating with your supplier to reseller and customer. PBX is an easy and cheap way to handle both internal and external affairs. Effective call distribution among staff, collecting customer information, and taking messages after hour are only a few options that PBX can offer.

Hosted PBX
Despite all of the above advantages, implementing your own system is not always worthy. Hiring an IT specialist to maintain your system can be highly pricey. In this case, the most effective solution would be to use a hosted PBX system, provided by a PBX carrier that will endure the maintenance for your specific requirements. An ideal PBX carrier would be user-friendly and include customizable features for your unique needs. You won’t need to pay for the PBX equipment, maintenance costs, installation, and upgrades. All you’ll need is your existing internet connection and some cables to connect the phones to computers.

No additional costs on the hardware
As mentioned before, you will not need to have special equipment. You can use the software on your personal computer or smartphone which allows for remote access when you’re not at your workplace. All you’ll need to do is to install the app. In case that you want to use a phone, you can use any VOIP compatible phone, for example, a popular Fanvil brand.

Integration with other communication systems
If you have an existing phone system that needs to be linked to the PBX it can also be arranged. To integrate this, the existing phone system will be linked with the Hosted solution via a SIP trunk. Moreover, this will allow the application of extra features such as a call to a specific agent, queue or voicemail when the SIP trunk is down.

If you are still in doubt that you need a Hosted PBX solution you can always sign up for a free trial to test the service and see how it addresses your unique needs!

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t38 FAX codec in FreePBX

I have received some requirements regarding Fax codec t38 on FreePBX with SIP service provider, spending some times to figure it out how should enable the t38 so here is the summarized outcome of my experience.

First of all we will need a trunk in FreePBX so here is the minimum parameter we need to have in the trunk setting:

username=SIP_USERNAME

type=peer

secret=SIP_PASSWORD

qualify=yes

nat=yes

insecure=port,invite

host=sip.t38fax.com

defaultexpiry=300

port=

————————————————————————–

Incoming -> Register String

 

SIP_USERNAME:SIP_PASSWORD@sip.t38fax.com:5080/SIP_USERNAME

 

————————————————————————–

Then you we need to enable the t38 Pass-Trough in the FreePBX setting >> Asterisk SIP Setting:

T38 Pass-Trough enable

————————————————————————–

Next we need to get access to CLI and modify the udptl_custom.conf file to look like below.

udptlstart=10000

udptlend=10999

udptlchecksums=no

T38FaxUdpEC=t38UDPRedundancy

udptlfecentries=3

udptlfecspan=3

use_even_ports=no

—————————————————————————

After all this setting you need to reload the FreePBX using below command in CLI

fwconsole reload

Vega Gateway setup with FreePBX High Availability

Recently I have setup a FreePBX high availability with Vega 200 and faced issue for the call, after spending sometime found out where is the issue. In FreePBX HA setting we will have one IP assigned to each machine and we have one floating IP which will assigned to the active machine automatically, by right w\you will need to pint any endpoint and gateway to the floating IP so it will always route the request to the active server. For voice gateways this scenario working for the inbound call however your gateway will faced issue for the outbound call as the active server when sending any call to the gateway will send the real IP address of the server but we have setup the trunk based on the floating IP so gateway will consider the call request received from unauthorized IP and will reject the call.

Based on my experience on this setup you will need to allow call from all the 3 IP address use for HA setup in the gateway configuration to resolve this issue.

Here is the setting for Sangoma Vega gateway with FreePBX HA setup.

If you have done the first configuration through the Quick Config for the floating IP to point the gateway to your FreePBX then you will need go to Expert Config >> SIP and edite your SIP Profile to look like this:

For this you need to add more SIP Proxy for the additional IP address of the HA setup. Simply add the SIP Proxy by clicking at the Add button and key in your IP and the port number and then save it. You can add as many as FreePBX node you have in your HA to cover all your node connectivity with the Vega gateway.

G.723.1 Codec Installation for FreePBX

G.723.1 is an audio codec for voice that compresses voice audio in 30 ms frames. An algorithmic look-ahead of 7.5 ms duration means that total algorithmic delay is 37.5 ms. Its official name is Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s. It is sometimes associated with a Truespeech trademark in coprocessors produced by DSP Group.
There are two bit rates at which G.723.1 can operate:

6.3 kbit/s (using 24 byte frames) using a MPC-MLQ algorithm (MOS 3.9)
5.3 kbit/s (using 20 byte frames) using an ACELP algorithm (MOS 3.62)

G.723.1 is mostly used in Voice over IP (VoIP) applications due to its low bandwidth requirement. Music or tones such as DTMF or fax tones cannot be transported reliably with this codec, and thus some other method such as G.711 or out-of-band methods should be used to transport these signals.

To install the G.723.1 codec for FreePBX you will need to download and save the module first by below command.
WGET http://asterisk.hosting.lv/bin/codec_g723-ast130-gcc4-glibc2.2-x86_64-core2-sse4.so

After download completed we need to rename and move the module into the Asterisk module folder by below command.
cp codec_g723-ast130-gcc4-glibc2.2-x86_64-core2-sse4.so /usr/lib64/asterisk/modules/codec_g723.so

Nex we will restart the sterisk in order to load the new module into Asterisk.
service asterisk restart

To confirm the installation you may check the Asterisk transcoding support table by executing below command.
asterisk -rx ‘core show translation’

If you see g723 above then both installed succesfully. Otherwise check the Asterisk logs.

Install Certified Asterisk V11 on Centos 7

For those are familiar with powerful Linux based IP-Telephone system Asterisk they may know there are two different types of Asterisk releases Standard, and Long Term Support.

Long Term Support release is fully supported for 4 years, while the standard releases are supported for a shorter period of time; one year of full support. Both version has an additional year of maintenance for security fixes.

Normally certified Asterisk releases have undergone additional testing and are made less frequently, generally identical to the Long Term Support release of Asterisk.

Users new to Asterisk are encouraged to download the latest Long Term Support release, with the guidance that Certified Asterisk branches based on those releases have the least churn.

Based on the recommendation for the stability we will go through the process of certified Asterisk as below.

We also recommend to use Centos as the Linux OS as it is based on the redhat and it is open source.

So first step after OS installation is updating your OS by yum command.

yum -y update

After fully update the OS we need to disable the SELINUX as it will interrupt the propare installtion of the Asterisk.

sed -i s/SELINUX=permissive/SELINUX=disabled/g /etc/selinux/config

Befor next step you required to restart you server in order to apply the new SELINUX configuration.

Next we need to install all the prerequirements for the Asterisk.

yum -y install bind-utils curl-devel doxygen gcc gcc-c++ gtk+ gtk+-devel gnutls-devel httpd kernel-xen kernel-xen-devel kernel kernel-devel kernel-smp-devel kernel-smp libxml2-devel libtermcap-devel libtool-ltdl libtool-ltdl-devel make mysql mysql-server mysql-devel mysql-server-devel mysql-connector-odbc net-snmp-devel neon-devel newt-devel ncurses-devel ntp openssl-devel openssl perl-suidperl php-mysql system-config-securitylevel screen texinfo unixODBC unixODBC-devel wget sqlite sqlite-devel

Then we change or location to src in order to download the source of Asterisk.

cd /usr/src

By below link we can download the source of Asterisk and save it in the src.

wget http://downloads.asterisk.org/pub/telephony/certified-asterisk/asterisk-certified-11.6-current.tar.gz

Next Step is untar the source file.

tar zxf asterisk-certified-11.6-current.tar.gz

enter to the Asterisk folder now to start the instalation.

cd asterisk-certified-11.6-cert18/

By executing below command you will see the Asterisk menu if required any additional module to enable or disable and then proceed to the full installation with the samle config files.

./configure && make menuselect && make && make install && make samples && make config

You should check the last stats of the installation process to make sure the installation was complete and successful. If the installation was successful you may enter to the Asterisk CLI by below command.

asterisk -r

if you face any error for the instillation you may need to double check the installation process or check if the error mention any missing requirement for the Asterisk installation.

ifconfig not exist in CentOS 7 Minimal Install

Perhaps one of the first things that you will notice right after your first login to the Centos 7 minimal installation is about the ifconfig command. This utility has been deprecated in Centos 7 and you have two option first there is alternative option to get and manipulate your IP address using the ip command:

ip addr sh

but if you still want to use ifconfig, as second choice you can use the “yum provides” command to figure out what packages provide certain commands or files you are used to using like below:

yum provides ifconfig

So in order to use the ifconfig command you just need to install the net-tools.

yum -y install net-tools

FreePBX Geometrical inbound call routing

I have a client asking to route the call based on the source coming from different location. We have a way for inbound call routing based on the prefixed of the caller ID in FreePBX. I have create a manual for how to setup this as below maybe useful for someone else:

We need to define the inbound route by going to the connectivity and inbound route then click on the new inbound route.

In the inbound route you will need to give it a name then in the CID need to mention the inbound caller ID in this format:

_”Country and state prefix”X.

Example:

Malaysia Kuala Lumpur: _603X.

Malaysia Pinang:              _604X.

Malaysia Ipoh                    _605X.

Then you need set the CID Priority Route to yes and select your destination for this route.

After saving this inbound and apply changes any call coming that start with this prefix will go to the defined destination.

This will use for distribute the call based on the source location to different agent for support or sales where different group will be responsible to handle each state or country request.

Centos V7 Time setting

You can found the current time setting by using “timedatectl” the output will be like below:

Local time: Fri 2017-06-30 16:37:54 MYT
Universal time: Fri 2017-06-30 08:37:54 UTC
RTC time: Fri 2017-06-30 08:37:54
Time zone: Asia/Kuala_Lumpur (MYT, +0800)
NTP enabled: yes
NTP synchronized: yes
RTC in local TZ: no
DST active: n/a

issue below command will show you the current time zone on the server setting.
timedatectl | grep Time

Sample output:

Time zone: Asia/Kuala_Lumpur (MYT, +0800)

in order to get the time zone name format for your location you can list all the available time zone by issue below command:
timedatectl list-timezones

After founding your local time zone name you can set the time zone by below command.
timedatectl set-timezone “Asia/Kuala_Lumpur”

At the end os setting to make sure your server has correct time zone setting issue below command.

timedatectl status