1 Online Direct

User Guide
Logging in, out and password reset

You’ll see a screen like this to log you in:

Logging In

Once logged in, you should be directed to the Home Dashboard

Password Reset

From the login page, click on FORGOT PASSWORD

Enter your username or Cloud.PBX number and click on SUBMIT

If authenticated correctly, an e-mail will be sent with a password reset link. 

Click on the hyperlink and follow the prompts to reset your password.

Logging Out

Permissions & Notifications

There are various permissions to be set and in addition you can allow the browser to serve notifications at the operating system level.

Setting permissions and notifications can be accesses from the dashboard.

Browser notifications for incomming calls, customising audio prefeerences, creating an avatar for your chat, or linking your google or O365 account, and much more.

Making Calls & Call Management

Making a call using Telviva One works just like any other telephony app:

Making a call

In Call

Call History

You can view a list of calls made or received using a date range. You can see the duration of the call and thespecific number or extension of the call. You can save the number as a contact.


You can view a list of contacts on the platform. There is a search function that helps you to filter your results. You can create a new contact and share compary wide or not.

Transferring Calls

To transfer a call using Online Direct 1, click on transfer, the original call will be placed on hold and offer options for transferring calls.


You can view voicemail notifications from the call menu.

Click on invoice to view read and unread messages. 

To listen to a voicemail, click the play button next to the message.

Once the voicemail is selected, you can move it to other folders.

Voicemail customisations is found under phone settings.



DND(Do Not Disturb) is a setting that allows you to stop your extension from receiving incoming calls.

Desktop Application

This allows you to download and install the desktop application to your PC/Laptop from the dashboard page.

If the Online Direct 1 Desktop application is not supported by your operating system permissions or if internal company Software Restriction Policies are enabled to block new applications being installed, the user needs to request assistance from their internal IT Department, as Online Direct cannot assist with company IT policies & configurations.


If you require additional assistance with the Online Direct 1 portal, desktop or mobile application, please contact us on 011 317 1800 alternatively, corporate.support@onlinedirect.co.za

Frequently Asked Questions

I can't log in, how do I retrieve my username & password?

Please see the section “Logging in, out & Password Reset” in the “Installation & Users guide” section above.

I can't hear the other party

Check your sound volume, mute settings and whether you are able to hear other sounds such as videos playing in the browser etc. If the sound is erratic, see the section about Requirements & Network above.

The other party can't hear me

Please check your microphone or headset first. You could use a tool such as sound recorder or your audio settings of your operating system to test sound input. If the sound is erratic, see the section about Requirements & Network above.

The call quality is poor

Your firewall should allow outgoing & incoming UDP to the public internet
We utilize WebSocket connections, so HTTPS / WebSocket / Secure RTP should
be allowed.

Local network conditions have the biggest impact on voice quality. Jitter, latency, and packet loss can be the biggest contributors to voice quality issues in any VoIP network.

High latency can substantially degrade a caller’s experience. While there will
always be some latency between the codec algorithm, the jitter buffer, and network
traversal, the goal is to keep this to a minimum. Callers typically start to notice the effect of latency once it breaches 250ms and find latency above ~600ms to be
nearly unusable. Here are some strategies to minimize latency on your network:

● Some lower bandwidth fixed internet connections can often have higher
latency. If possible, upgrade your internet connectivity.
● Stick to high-bandwidth connections. Mobile networks such as LTE (mobile
4G Data) can often have high latency

Packet loss, most frequently jitter-induced packet loss, can greatly impact your
VoIP call quality. Wi-Fi can be particularly bad for creating jitter. Here are some
strategies to minimize jitter on your network:

● Reduce packet conflicts on Wi-Fi by reducing the number of devices operating
on the same channel.
● Avoid large data file transfers concurrently with voice over the same Wi-Fi environment.
● Avoid buffer bloat, which can result in high latency, and bursts of jitter. We
recommend ensuring your router is configured with the low buffer size, as high
jitter cannot be masked by a buffer without introducing artificial delay, and often choppy audio. Note: Not all routers allow for configuring buffer sizes, but some routers ship with defaults that are not optimized for real-time VoIP networks. Open-source routers, enterprise-grade routers, and gamer-oriented routers are good candidates for providing the right configuration options and defaults.

If you have addressed the above issues and continue to have jitter related impact on
your voice quality, you may consider configuring your router with QoS rules to
prioritize traffic on the above media UDP ports. Given the large range of UDP
ports, you should only do this with prior consideration to what other traffic may be
flowing in that port range.

Call quality
By following this guide, you can significantly improve the quality of service for wireless voice applications and reduce or eliminate dropped calls, choppy speech,
fuzzy speech, buzzing, echoing, long pauses, one-way audio, and issues while
roaming between access points.

Network MOS – The Network Mean Opinion Score (MOS) is the network’s
impact on the listening quality of the VoIP conversation. The score ranges from
1 to 5, with 1 being the poorest quality and 5 being the highest quality.
● Packet Loss Rate – The packet loss rate is the percent of packets that are lost
during transmission.
● Interarrival Jitter – Interarrival jitter measures the variation in arrival times of
packets being received in milliseconds (ms)

Below is a summary of the best practices to provide the best voice quality over
Perform a pre-install RF survey for overlapping 5 GHz voice-quality coverage with -67 dB signal strength in all areas. (Use Wifi Analyzer App)
If possible, create a new SSID dedicated to your voice over IP devices.
● Set Authentication type to ‘Pre-shared key with WPA2’
● Set WPA encryption mode to ‘WPA2 only’
● Enable ‘5 GHz band only’
Enable ‘Traffic shaping’ on the SSID to prioritize all voice traffic
SIP 5060 UDP / TCP – RTP 10000-20000 UDP – internal Network / UDP 65550 if
Vibe goes via Firewall
● network
● network
● network
● network
Set DSCP to ’46 (EF – Expedited Forwarding, Voice)’ for RTP

Video Meeting

Note the use of WebSockets – both to collect “events” and for the webrtc for calls. Some customers may have a web proxy running – some older web proxy software is not compatible with WebSockets even though they are standard.
● address, TCP port 443: Used for access to the Online Direct 1
backend services – straight HTTP requests used, but also wss (secure websocket) connection for PBX events from Telviva.
● rtcproxies (, to follow):
● tcp port 4433: WebRTC connections (this is an encrypted websocket
connection carrying SIP packets to and from JSSIP on the client system).
● tcp port 3478: STUN and TURN service, a requirement for WebRTC.

1. TCP port 443 is the default port for secure HTTP. It would be very unusual for
that to be blocked; if it were blocked most websites wouldn’t work.
2. TCP port 4433 is a fairly commonly used secondary port for secure HTTP. But
more likely that it is blocked. It might have to be unblocked on the firewall.
3. TCP port 3478 is for STUN/TURN, a requirement for all webrtc and used by
modern SIP phones to help deal with NAT. It should be opened.
A customer using a WEB PROXY would need to check that it is compatible with
the use of WebSockets. Websockets are a web/internet standard
(https://tools.ietf.org/html/rfc6455) that should be supported.

How much data does Online Direct 1 VoIP use?

VoIP calls use under 1Mb per minute.
With 1Gb you should be able to make over 16.5 hours worth of calls.


I couldn't find my answer here, how do I contact support?

You are welcome to contact Online Direct Support on the following channels:

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