Help Topics

Hosted PBX phone system is a service provided online via internet. It allows to connect special internet phones (VoIP phones) and make/receive phone calls. Hosted PBX also allows configuration of call-flow features to route calls using predefined rules to the required user/extension. SIPPALS provides an extensive feature set free of charge, including popular features like time of day call routing, voicemail to email, call queues, follow me, voice menus and much more.

Cloud PBX, IP telecommunications, Virtual office phone system. There are plenty of names for this technology in the telecommunications industry. PBX Stand for “Private Branch Exchange” and in the past allowed communications within one building or office. Modern telephony service use internet to transmit voice and PBX can be hosted in the data centre or in the cloud. It is basically a set of equipment and software that allows advanced call routing functionality.

PBX stands for Private Branch eXchange. In the past, due to a limited number of phone lines being connected to a building, the primary function of PBX was to allow internal calls between system users and for connection of more phones to the system than lines available. Lines were effectively shared among users via PBX. Modern technology nowadays using VoIP does not require traditional PSTN lines and uses data links instead. PBX is still widely used, although its primary function is now to improve call handling and user performance via advanced call routing features. Modern cloud PBX is effectively software that controls digital voice data.

Hosted PBX is based on Voice over IP. In simple terms, your voice is encoded by the phone and transmitted via the internet to the server with cloud PBX Software. Hosted PBX server in turn can perform various tasks with the routing of your call and then pass it to upstream carriers for connection to the dialled destination. Most modern carriers connect to each other using IP data links. This allows for better quality of sound, more security and reliability.

VoIP (Voice over IP) requires data connection between your premises and our servers. In most cases it would be public internet. However you only need a tiny amount of bandwidth – connection speed in order to use VoIP. Each channel requires 80Kbps (0.08 Mbps), or using compressed codec even smaller – 0.03Mbps. Quality of VoIP will highly depend on the reliability of your internet connection. It does not have to be very fast, but it must be reliable and stable. We also recommend using dedicated internet link for setups with over 5 phones.

We have multiple solutions that can make sure you can continue to receive calls even when your internet goes down. You can have calls diverted to your mobile phone if you are offline, this way you can continue to receive calls without any internet.

Simply speaking – it is same as traditional phone lines. We use G711 voice codec as standard. Connection will be crystal clear and same as normal PSTN services. Also, unlike many other services, all our equipment is hosted locally, meaning that there will be no delays in voice transmission.

Our systems are optimized for reliability, not price. We guarantee 99.9% SLA backed uptime on all our systems. All parts of the system are fully redundant, meaning that there is no single point of failure. This is not just words.

Hosted PBX is based on Voice over IP. In simple terms, your voice is encoded by the phone and transmitted via the internet to the server with cloud PBX Software. Hosted PBX server in turn can perform various tasks with the routing of your call and then pass it to upstream carriers for connection to the dialled destination. Most modern carriers connect to each other using IP data links. This allows for better quality of sound, more security and reliability.

Your older handset can be used with VoIP via special device called ATA (Analog telephone adapter). Adapter will convert analog signals sent by your phone into IP traffic that can be transmitted over the internet. However, we recommend using special phones like Sangoma or Fanvil for the full experience of VoIP as it gives you an additional range of great features and greatly improves the quality of calls.

VoIP technology allows small business to benefit from call routing features previously available exclusively to large corporations due to high price associated with complex onsite PBX systems. Today technologies allow small business owners to greatly improve their call handling by using features like multiple simultaneous calls on the same phone number, call distribution using various rules, ability to answer calls from anywhere you can get access to internet. It is also significantly cheaper compared to older PSTN or ISDN phone lines.

VoIP (Voice over IP) requires data connection between your premises and our servers. In most cases it would be public internet. However you only need a tiny amount of bandwidth – connection speed in order to use VoIP. Each channel requires 80Kbps (0.08 Mbps), or using compressed codec even smaller – 0.03Mbps. Quality of VoIP will highly depend on the reliability of your internet connection. It does not have to be very fast, but it must be reliable and stable. We also recommend using dedicated internet link for setups with over 5 phones.

Number of Concurrent Calls During Busiest Hours

X

100 Kilobits per Second

=

Bandwidth in Kilobits per Second Required for SIP Trunking

We have multiple solutions that can make sure you can continue to receive calls even when your internet goes down. You can have calls diverted to your mobile phone if you are offline, this way you can continue to receive calls without any internet.
Oubound calls would be interupted till your internet service back to normal.

Simply speaking – it is same as traditional phone lines. We use G711 voice codec as standard. Connection will be crystal clear and same as normal PSTN services. Also, unlike many other services, all our equipment is hosted locally, meaning that there will be no delays in voice transmission.

Our systems are optimized for reliability, not price. We guarantee 99.9% SLA backed uptime on all our systems. All parts of the system are fully redundant, meaning that there is no single point of failure. This is not just words.

• Subscription upgrade is possible any time and the process will take one business day to be completed. New call rate would be effective from the upgrade plan completed.
• Subscription downgrade would be possible any time and new plan would be applied on next day of subsequent month.

• please login to user portal and from left menu select SMS.
• on SMS panel click on New

• on next windows fill up the required field, all field are compulsory to be filled

• Number: is your SMS destination with country code, do not include + or 00 in front
• SMS: is the text message which you are going to send, maximum charachter which can be send is 140 charachters including space
• From: please put your name/company name or brand here. Please note you are not allowed to put international brands (Like Apple, Google, Paypal, etc) name for this field otherwise your account would be charged RM10 per such SMS and if repeated your account would be suspended. From length yould be between 4 to 10 charachters.
• Once all field being filled click on Save to send your SMS

• Yourusername:: Use your SIPPALS portal username
• MD5PASS: we need encrypted password of your SIPPALS portal, you may convert your password to MD5 using this website: https://www.md5hashgenerator.com/
• RECEIVERNUMBER: key in your SMS receiver number including country code (without + or 00)
• YOURTEXT: your SMS text which is maximum 140 characters
• FROM: key in 4~10 character of your company/brand name as sender, using international names like Paypal, Apple, etc will cause penalty and account termination