Question: Using the dialler system for outgoing calls it starts to dial out on the headset but then stops after a couple of seconds - why?
Answer: This could be a browser issue – clear the cache and cookies on the browser. Make sure you are using the most up to date version of the browser and ensure you are not using Internet Explorer as this is not supported.
Question: Using my headset with the dialler, the headset is making a ringing noise - but there is no outbound call or incoming call - the dialler is blank.
Answer: Please clear the cookies and cache on your browser, then reload the CRM and see if you can successfully make and receive a call. Clearing the cookies and cache will normally resolve such issues.
Question: We are unable to make outbound calls - when we dial a number via the dialler system it says 'Finished'
Answer: Please try clearing the cache on your browser and re-loading the system again. Also try dialling *60 and *43 to see what happens.
If this issue persists please check which VoIP engine you are running as if using the Web RTC which isn't compatible for all routers you may need to switch to the VoIP NS Engine.
You will need to ensure you are on this via the Settings within the dialler system, select: Settings - Advanced Settings - NS Engine. This will then resolve the issue for them.
To activate the NS Engine, go to ";Settings"; /";Advanced Settings"; / ";VoIP Engine"; and select ";NS Engine"; from the
Question: Using the dialler to make outbound calls - when I click on a number the outgoing call is terminated straight away - a message pops up on the dialler to say 'call terminated' then it goes back to 'registered' and then back to the keypad.
Answer: Please try logging out and logging back in again, it should then work OK. If not please clear cookies and cache from your browser and login again.
Question: I have an error message 'Register - failed forbidden' on the dialler but it was working OK yesterday, why?
Answer: You may need to change the VOIP engine you are using to run the dialler from WebRTC to NS engine. To do this please follow the guidance below:
To watch a video on how to do this please click here. Alternatively, please see the step by step guide below:
On the Dialler select the menu in the top right-hand corner (it has 3 horizontal lines)
Click on Settings
Click on Settings…Show Settings
Show Advanced Settings
SIP Settings – VoIP Engine
Make sure that NS (service plugin) is selected as shown below:
You will then need to download the web phone service file – if you get a blue message box – click on ‘read more’ and then you will see the option to ‘Run’ the download
When the download has completed click OK
You may need to logout and clear your browsers cache before logging back in.
You will then see that the dialler system is back online and ready to use again.
Alternatively, if you are using the NS engine, you may need to switch over to WebRTC - you can do this following the same instructions as above, just select WebRTC and click 'OK'. You will then need to login again for the change to take effect.
Question: Poor quality call
Answer: Poor Internet Connection- because VoIP relies on your Internet connection to deliver and receive the voice packets that make up a call, the speed of your Internet connection is crucial to call quality.
Internet connections with low-bandwidth limit your VoIP call quality. If you are using an insufficient broadband connection with limited connection speed then you can expect your VoIP call quality to be less than spectacular.
Saturated Internet Connection
Think of Internet saturation as the over-sharing of available bandwidth. Saturation reduces the bandwidth available for each call and can lead to reduced call quality. The saturation can occur on your own laptop, on your LAN, or at your router, whether you are at the office or working at home.
Even if you are not currently using a specific application, like Microsoft Office or Adobe Photoshop, you are still using your bandwidth and therefore could be affecting your VoIP call quality. In fact, many laptop services run in the background and use your Internet connection.
How many people in your household or at your office are using your Internet connection? Do you know how many applications or services they have running? Each user probably has as many services using the Internet as you do. The more users you have in your work group, company, or household, the more saturated your internet connection will be.
If you think about how many people live in your neighbourhood or work in your office building - if your Internet travels through cable like fiber optic cable or copper, then you are all sharing the same Internet connection.
Your Broadband Provider is Unreliable
There are many Internet Service Providers (ISPs) to choose from with each provider claiming to have high-speed connections at a low-cost, when in reality many of them do not have the capacity to deliver such services.
You can use VoIP connection testing to assess what you can expect from an ISP - please click here to see how you can search for particular broadband providers in your area.
If you are still not sure that you have found a reliable Internet service provider, get a trial and test your connection with a reliable service such as www.speedtest.net for overall network performance to check factors affecting your VoIP call quality.
For really good VoIP call quality, you should see jitter at about 1 ms (milliseconds) or 2 ms, or less, and packet loss should be 0.0%.
For “acceptable” call quality, your jitter can be as large as 7 ms and packet loss can reach 1%.
You can also do a trace route command which examines the traffic on your broadband line - please click here to see how to do this and send us the results back so that we can advise further.
Outdated routers, firewalls, and cable modems can also contribute to poor VoIP call quality.
If you are experiencing poor call quality, you should either update your router software or replace with new equipment.
You can examine each network element between your laptop or VoIP device and the internet to isolate older components that can be updated or replaced.
Please click here to read more about system requirements for VoIP and router information to help ensure good quality VoIP.
Checklist for Call Quality Issues:
- If call quality is affected (such as voice breaking up, crackling on the line, voice fading in and out etc) please can you confirm is it inbound or outbound calls? Or are both affected? Please confirm.
- Is it call connectivity issues - for instance is the call (incoming or outgoing) connecting - can you hear the person on the other end of the line?
- Please make sure that call recording is set up on your VoIP line(s) that are affected, please click here to see how to set this up.
- Can you play back the call recordings - are they affected or can you hear the call recording clearly? Please confirm
- Who is your broadband ISP provider?
- Are you aware of any faults with your ISP?
- What router do you use - please provide make and model?
- Has SIP ALG been disabled on your router?
- Are you on a Static or Dynamic IP address? If on a Static IP address is DHCP enabled on your router?
- Do you have a firewall that may be blocking your VoIP traffic? Make sure that ports 5060 and 5065 are open.
- Do you share your broadband line for both web and VoIP use?
- If you do not have your own dedicated broadband connection for VoIP then you need to ensure that you use a router with a quality of service (QoS) setting to prioritise voice traffic over regular traffic as this is an absolute must. Without QoS you will encounter poor quality telephone calls regularly.
- Please do a ping to your VoIP IP address - please click here to see how to do this and then get in touch with our Support Desk with the results so that we can take a look at packet loss and jitter results.
- Please do a Traceroute command click here to see how and send us the results back
- What VoIP phone equipment are you using, please provide make and model of handsets and headsets or name of softphone client?