QoS for ViBE

Vibe's port is exclusively UDP 65500, similarly, with PBX QoS. Destinations are:

 Please Note: The port is exclusively 65500 bidirectional.

Checklist for Network 

Please Note: Always check physical media first.

1. Check that the following settings are disabled on the router/firewall:
1. SCTP
2. DCCP
3. UDP lite
4. SIP ALG / SIP Direct media
5. IGMP Snooping
2. Sharing voice and data? If yes, data (TCP) takes priority over voice (UDP) and will cause a degradation of quality.
3. Was the correct QoS information used from Dropbox?
4. Is the traffic bi-directional?
5. Has a different network cable been tried?

Why is there No Connection Between the Phone and the Gateway of the LAN? 

• Check that cables are inserted correctly.

• Verify that the network cable has got an internet connection active on it. ( Test this by trying to plug a cable that is working on another phone into the phone, giving the error)

• Ensure that the phone has the correct network IP address

Reasons why the phone is not Registered with possible corrections.

1. SIP password that is incorrect (Recommended to copy/paste) or that the PBX server is not correct when setting up a phone.
2. If all the details have been copied & Pasted, look at the phone settings. Usually, start with registering the phone on Account 2.
3. Ensure the Network Cable and Network Port are both active and give out DHCP (unless network setup is only allowing Static IP addresses. In this case get an IP Address assigned to the device and set it up statically).
4. Ensure the cables are in working order as it could be that the network cable is faulty.
5. Ensure that the network cable plugged into the phone is connected to the “Internet “port on the phone.
6. The Protocol should always be UDP.
7. Change the local SIP Port, found in Account Settings under the advanced tab or under the settings tab in the SIP category. The SIP Port will always be 5060 by default. You can change the SIPPort to something else between 5060 and 65500.
8. Try moving the phone to a different network point where there is a working device. It could be that the network point is causing an issue.
9. Try and register the PBX domain directly on the IP address as there could be a DNS translation error. (Gateway router is not translating the PBX to the relative IP). This issue could be resolved by troubleshooting your network/gateway router.
10. If the above steps do not work, you could confirm if there is are VLANs set in place on the customer’s network as well, which the phones need to be on.
11. In rare cases, the customer’s network can have issues as well, such as DNS issues. If you change the phone’s DNS to the following, the phone should register:
1. Primary: 8.8.8.8
2. Secondary: 8.8.4.4
   

If the phone does register after the changes have been made, the network would need to be looked into from the customer’s side.

Completing the above steps and the phone doesn’t register, the customer would need to look into their network setup and check if there is any conflict on their end.

Please Note: Make sure that SIP ALG is disabled on the WAN routers. When using a Mikrotik specifically, it’s called SIP Direct Media. With Cyberoam, the SIP ALG setting can only be disabled from the CLI. SIP ALG was designed to help VoIP, but it causes extensions to de-register. Disable IGMP Snooping.

Why a User is Unable to Call an Extension.

1. When calling the extension and it goes directly to voicemail, it is most likely on DND or offline. Disable DND and the call will go through. DND rejects ANY call sent to the phone.
2. The agent could be paused In the Queue (Enterprise PBX only offers queues). However, this would only allow the extension not to receive external calls from the queue.
3. For new customers: Confirm with production that the account has been moved on from installation.
4. In extremely rare cases concerning cordless phones, the number assignment is not in order.

Why will a user be unable to call out but receive calls?

1. Should there be a firewall on the network, it could be that the traffic is allowed to go out but not come back in. If firewall rules are implemented ensure its bidirectional. (Allowed in and out)
2. Registration dropped from the network, rebooting the device could resolve the issue.
3. Reaching the extensions budget limit.
4. In extremely rare cases concerning cordless phones, the number assignment is not in order.
   

Reasons why the ViBE Router is not coming Online.

1. On some occasions, the ViBE will not come online because the password still remains “TenantName”. This password would need to be replaced with the PBX’s Tenant Name which can be retrieved from the TMS.
2. The Primary connection is not set correctly.
3. By default, Port 2 is set to have a static IP 192.168.220.2 which connects to the primary internet connection (ADSL/LTE/Fiber) which will be configured with a gateway of 192.168.220.1. If you have a dedicated internet connection or are going to share your internet between voice & data, plug that into port 3 of the ViBE and either remove the route on the ViBE, or assign a reserved IP to port 3 of the ViBE and update the route.
4. Ensure the license from the supplier is purchased  and that the CPE is activated.
5. Ensure the details sent to Euphoria are correct such as the Mac address and password.
6. Check if the ViBe config file loaded on the vibe is corresponding to the server where the Euphoria agent added it to e.g. ViBE1, ViBE2, etc
7. Cables swapped.

Why are there Call quality issues

1. This is mostly experienced when a customer is sharing the internet with voice and data. In this case, a dedicated Internet Connection for Voice is needed. 
2. When using Fibre, it would be recommended to get a minimum of 2MB dedicated for voice (Get in touch with your ISP to do this on their end). 
3. The ViBE can be set in Rain Mode on the CPE, but on the headend the ViBE will be set to Failover (Vice versa). Call/mail Euphoria Support to confirm the server set-up.
4. When there are call quality issues like crackling, a loud buzzing noise etc., Check that the curly cord is plugged into the correct port.
1. Try calling on loudspeaker.
2. Try changing curly cord.
3. Try changing the hand piece.
4. Check firmware of device.
5. Factory reset the phone.
5. If the caller says that they can’t hear you, experience one-way audio, that you sound like you are underwater or disappear etc., this is most probably a Firewall issue. Ensure that the correct protocols are followed , sip ports and UDP ports are allowed through the firewall, as well as from the ISP side. (Ports and IP’s are provided on the Reseller Dropbox in the Network Configuration folder.
6. When there are call quality issues, confirm what kind of phone it is as well. Is it a desk phone/cordless phone/cell phone app/Softphone etc.? With this being said, if it is a cordless phone, make sure you are not out of range from the base or if there is anything else near it that can cause interference.
7. If using a softphone: 
1. Try a different internet browser like Google canary.
2. Ensure the headphones input and output is working on the pc.
3. Register the extension on a different internet connection to establish if the issue is connectivity related or pc/hardware related.

If a Ported Number cannot be Reached after it has been Ported to Euphoria.

1. Is this from a certain Network or from the Euphoria network, including phones on the Euphoria Network.
2. When porting takes place it's generally between 5pm and midnight as other providers need to update the routing tables on their side. Some calls might come through and some might not, this is part of the porting proves unless there has been a number move scheduled. 
3. Ensure that the routing on the Euphoria’s side is correct when this happens to all phones. When the message  “I’m sorry, the number you have dialed cannot be found on the Euphoria Network”, is heard, it means the number has been ported, but not been added to the Account yet.
4. If the routing is set up on Euphoria’s network, get the latest example of a call that happened within, at most, a 3 hour time frame and escalate this to Euphoria who will find out what the issue is. On some occasions, the routing has not been updated on Vodacom, Cellc, MTN etc side (The relevant network provider needs to sort this out on their side). Send the example to Euphoria Support to investigate and escalate it to the relevant network provider if needed. Please see example below:
1. Source call (Number that you called from)
2. Destination call (Number that was called to)
3. Time that the call was made Error message received
   Please do not contact the network provider directly, all correspondence with relevant network providers MUST be done through Euphoria Telecom.

Bugs and Fixes

Unidata Wireless phone- Phones not registering

Step 1: Navigate to the Basic Call tab.

Step 2: Uncheck the "Remove dash on alias" option

Grandstream GXV3240 Issue - Call Quality

Issue: The call quality is bad when receiving but not when making a call.

Fix: Codec Negotiation Priority - Set it to the caller and not the callee. 

Snom Soundpoint Reset

Snom Soundpoints have another way to reset if getting the password is not an option, below are a few methods that can be tried:

1. For SoundPoint IP 301, 501, 550, 600, 601, and 650 press and hold the 4, 6, 8, star on the dial pad at the same time. 
2. For SoundStation IP 4000, 500, and 6000: 6, 8 and star dial pad keys. 
3. For VVX-Series phones press and hold the dial pad keys 1, 3, 5 simultaneously during the Updater process until the password prompt appears
   

Reset Method 1:

The first method to reset the phone is while it is booting up. When the countdown screen is visible whilst booting up the phone, press and hold down 6, 8, and *. Hold these three keys until the countdown ends and a password is requested. By default, this password is 456. Enter the password to reset the phone to factory defaults.

Reset Method 2:

This can also be done via the phone’s menu if the phone has already booted up. Proceed to Menu / Settings / Advanced and enter the admin password (default: 456). Choose the appropriate option and the phone will boot to default settings on the next boot.

Forgot Admin Password of the SoundStation IP 6000 conference phone? 

When booting the phone to the password screen, enter the MAC address of the phone in the password field.

The MAC address can be found on the bottom of the phone. 

Enter the MAC address as all lowercase letters with no colons in-between sets.

Sip Vicious / Phantom calls / Ghost Calls

• These phantom calls are also known as SIPvicious are a type of SIP denial of service attack. This happens when the phone sits on a live IP address.

 A bot which is doing port scans on random public IP addresses will find an open port. The bot will then send SIP signals to this device on a general SIP port and the phone will ring. The caller ID will say "100" or "1000" and ring the phone constantly,usually at night, and when they pick it up there is no one there. These are so called ghost calls, it will disturb and confuse the phone user. No record will be found of these calls in the CDR's on your PBX as it is just a signal sent to the IP and it is not an actual call.

1. Ways to fix this:
1. Look at NAT rules on your router, make sure it is masquerading your internal IP's so they are not accessible from outside of your network.
2. Up the firewall level on your router.
3. Deploy a ViBE router (This is essentially a VPN and can't have packets injected)
4. Try changing the port on your phone, try something higher like 5090. They only scan the really common ports such as 5060, 5061, 5062 etc. You could also check the following:
   

1. IP Call feature allow theSIP phone to make and receive calls by IP address (Without SIP account)
2. Change by WEB UI path: Features->General Information-> Allow IP Call
   

1. *Change by Auto provisioning parameter
1. # 0-disable; 1-enable; default value is 1 (Enable)features.direct_ip_call_enable =
   

2. Enable SIP trust feature:

1. SIP Trust feature allow the phone to only accept the message from the trusted server for specific account.
2. Change by web UI path: (only V73 or higher firmware support. for W52p/W56p the firm ware should be V80 or above, if the W52p firmware is V73 or below, should change the setting via auto provision): 
   

Accept Features->General Information-> SIP Trust Server Only

1. * Change by auto provisioning parameter
1. :# 0-disable; 1-enable; default value is0 (Disable)account.X.sip_trust_ctrl =

How to change the local SIP Ports on different Devices.

Zoiper:

Desktop:

• Open 'Settings', then  'Preferences', then 'Advanced' (the Gear or Wheel button), then 'Network'.

• The 'Port' setting Under 'SIP Options' is the Local SIP Port.

• The 'Port' setting under 'RTP Options' is the starting Local RTP Port.

• Disable the options to 'Open random ports'.

Apple iOS:

• Click on 'Settings', then “'Advanced' to change Zoiper's Local SIP and RTP Port settings.

• Disable the options to use Random Ports.

Android:

• Find the Local SIP and Starting Local RTP Port settings under 'Config', then navigate to  'Connectivity', then  'Listening Ports'.

• Disable the options to use Random Ports.

Aastra:

• The Local SIP and RTP ports are under 'Advanced Settings' on the 'Global SIP' page. 

• A different Local SIP Port value should be chosen for each active account.

Bria:

Apple iOS:

• From the Dialpad click 'Settings', then 'Advanced Settings'.

• Under 'Media Options' the SIP Port Range and 'RTP Port Range for Audio' can be chosen.

• The Start and End SIP Ports should be the same value (don't use a range of SIP ports).

Android:

• From the dial pad click the Gear icon to access Bria's settings.

• Tap on 'Advanced Settings' to change Bria's 'SIP Port' and 'RTP Port Audio' settings.

Bria Desktop:

• Open Your sipgate account Settings and click on the 'Topology' tab to change the softphone's Local SIP ("Signaling") port and RTP ports.

Cisco/Linkys SPA:

• Each 'EXT' or 'Line' under 'SIP Settings' has a 'SIP Port' setting.

• The RTP ports will be found under 'Voice' --> 'SIP' --> 'RTP Parameters'.

• A different Local SIP Port value should be chosen for each active Ext or Line.

CSIP Simple:

• From the Menu below the Dialpad select 'Settings', then 'Network' to change CSIP local ports.

• The Local SIP Port is called the 'UDP Port - port number to bind locally'.

• With the Expert set up Wizard selected, each VoIP account can also be allocated a different "Port for RTP ports range start".

• Each VoIP account can be assigned a different Local RTP Port.

DrayTek:

• To change a DrayTek's Local SIP Port see the DrayTek Help article.

• A different Local SIP Port value should be chosen for each active Index.

Gigaset:

• The local ports are located under 'Telephony', then 'Advanced', then 'Listen Ports for VoIP Connections'.

• Do not enable the options to use Random Ports.

Grandstream:

• The 'Local SIP Port' is located in the 'Account' menus and the 'Local RTP Port' will be in the 'General' or 'Advanced Settings' menu.

• Each Line or Account can be allocated a different Local SIP Port.

• A different Local SIP Port value should be chosen for each active line.

Media5-fone:

• The Local SIP Port and RTP ports settings are found under 'Settings', then 'Advanced', then 'Network Options'.

• The start and end SIP Port value will be the same (one SIP port).

Panasonic:

• Each Line can be allocated a different Local SIP Port ("SIP Source Port") under "VoIP" under "SIP Settings", then 'Line x'.

• The RTP port settings will be found in the "VoIP Settings" menu.

PhonerLite:

• Open 'Configuration', then Network, then Local Port.

• The RTP port used by PhonerLite will be set to two ports higher than the chosen Local (SIP) Port.

Snom:

• Under Advanced

•  SIP/RTP the Local SIP Port is called the 'Network Identity Port' 

• The Local RTP Ports are called the 'Dynamic RTP Port Start/End'

 X-Lite:

• Open sipgate account Settings and click on the 'Topology' tab to change the softphone's Local SIP ("Signaling") port and RTP ports.

Troubleshooting - External Client (EC) 1.7.4. Overview

This document is created to assist a user in resolving the External Client issues when the user is not the administrator. This document also indicates what is needed in the case of network or firewall restrictions.

System Requirements

The machine recommended specifications:

• Operating system: Windows10 (x86 or x64)

• Ram: 4GB

Other requirements:

• The External Client also relies on .Net Framework 4.6 (Included in the EC setup)
  

Proxy

A WebSocket is being used as a communication channel between the browser and the External Client.

The External Client uses the alias localhost.euphoria.co.za as an identifier for the WebSocket connection on the local machine. 

If you are running a proxy server, External Client will require localhost.euphoria.co.za to be added to the exceptions in Windows proxy settings. Example below:

 

General

Connection

• The External Client requires ICMP  access upon startup in order to detect if the user has access to certain URLs. It will do this by running a ping test to Amazon S3 and Google.
  

Ports

The external client requires certain ports to be accessible in order to function, these summarily are:

• TCP 65501 which facilitates communication between the browser and the external client.

• EC uses UDP port 5060 to listen to SIP traffic. 

• EC will pick dynamic outbound UDP ports from the range between 10,000 - 20,000 to be used for RTP (Real-time Transport Protocol) and  RTCP (Real-time Transport Control  Protocol) which is used for audio streaming.

• PortSIP uses random loopback ports which do not require to be opened in the system firewall, these are for internal application use.
  

Site Access

The External Client will need to be able to access certain sites to be able to function correctly, these are:

• PBX server name (including the domain name), in order to handle the real-time communication, firewall rules permitting traffic to the server may be required.

• Amazon AWS used to download the certificate needed to secure the local port connection communication between EC and the user’s browser.
  

Local Certificate Store

The External Client needs to have read and write access to the Windows Current User Certificate Store. This will enable it to download the certificate(mentioned above) to the store folder and read it from there when needed.

The User certificate store path can be found in the following location: 

• {system drive}:\Users\{Username}\AppData\Roaming\Microsoft\Crypto

Updates

Auto Update 

The External Client requires administrator rights in order to auto-update, if it does not have these, it will need to be done manually through the administrator by running the External Client Updater from the system start menu or the application folder.

Issues

Audio Issues

Since the External Client will use the system communication audio devices, any troubleshooting necessary should be done through the system troubleshooter or on the actual device itself.

The External Client should be allowed to access the device microphone and that can be done through the system privacy settings.

• Speaker device: 16 bit, 48000Hz (DVD Quality)

• Microphone device: 2 channel, 16 bit, 48000Hz (DVD Quality)

User Conflict Issues

The External Client currently supports a single user at a time and it can only interact with a single browser page that supports it. Having more than a single page interacting with External Client will interrupt the main connection between the browser and External Client.

This can be resolved by keeping a single browser page open with the phone (The page may need to be refreshed) and then resetting the External Client (right-click the tray icon in the taskbar and reset).

Local Port Connection Interruption

Due to the External Client creating a local port connection with the browser, the connection can be interrupted by external factors like the system going to “sleep” mode or by the browser itself.
This can be resolved by resetting the External Client (right-click the tray icon in the taskbar and reset).

Other Issues

To detect any other issue, run the Euphoria External Client Console from the user system start menu and you should be able to see the logs and errors in detail. Please note that this can not be run at the same time as the normal client as it will conflict.

WebRTC Trouble Shooting

Getting started with Euphoria WebRTC

WebRTC (Web Real-Time Communication) is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, and P2P file sharing without the need of either internal or external plugins.

Your Euphoria PBX that you are currently using to make and receive calls is capable of making and receiving calls via Secure WebSockets, a module of WebRTC. You will need to familiarize your self with WebRTC and Secure WebSockets to continue.

Technical Specs and Requirements

Euphoria Allows for Secure WebSocket connections on port 443. If you normally communicate with your pbx at pbx4.euphoria.co.za on the regular SIP port of 5060, you can also establish a Secure WebSocket connection on port 443. e.g.  wss://pbx4.euphoria.co.za:443. Connections using Secure WebSockets are protected via TLS.

Euphoria does not support non-secure WebSocket connections.

RTP media is transmitted over UDP ports 30000-40000. (With a non WebSocket connections, like 5060, RTP would normally use port 10000-20000). Additionally, RTP media offered by your Web Browser is normally secured with SAVPF/TLS.

The Audio codec used by most modern browsers will be OPUS. Euphoria supports opus audio codec, however you will need to contact your support or sales consultant to have this option enabled. (Browsers will also offer ulaw and alaw - these two codecs are not recommended, and also have to be enabled by contacting Euphoria.)

The Video codec used by Chrome will be VP8/9 and Mozilla will be H264. Trans-coding video is not possible and as a result, video will not be possible via this method. An alternate video conferencing solution is available if required.

A STUN server is also available at the same pbx url. e.g., if you register with pbx3.euphoria.co.za, then the stun server will be stun:pbx3.euphoria.co.za

Building a browser based phone

There are two popular JavaScript libraries that enable full telephone functionality like call set-up, dtmf, call answer etc.

Option 1) SIPjs:  https://sipjs.com

SIPjs is an easy to use fast to get going script library, it can handle most calling features. It is recommended to use SIPjs JavaScript library to build a browser based phone, for simple use case scenarios. (At the time of writing, hold and un-hold was not possible).

Option 2) JsSIP:  http://jssip.net/

JsSIP is a more powerful script library but can be more complicated to setup. It is recommended to use JsSIP JavaScript library to build a more feature rich scenarios.

Downloads and Tool issues in TMS

Euphoria Advanced Softphone 

How to Authenticate the Euphoria Advanced Softphone or Agent Manager.

The Euphoria Agent Manager/Softphone app  makes use of the tms.euphoria.co.za and api.euphoria.co.za to authenticate.

These destinations should be unblocked on your network. The TMS and API both have two destinations, the DNS works in round robin fashion. It's wise to have both destinations for both services open as it could resolve to either.

tms.euphoria.co.za operates on port 443 TCP

Resolution:

tms-a.euphoria.co.za

41.79.37.34

tms-b.euphoria.co.za

41.79.37.44

api.euphoria.co.za operates on port 443 TCP

Resolution:

api-a.euphoria.co.za

41.79.37.39

api-b.euphoria.co.za

41.79.37.46

Why can't I make calls from my Euphoria Advanced Softphone (Agent Manager)?

I have an Internet Connection, but the Softphone Shows “Initialized’

A SIP account needs to be loaded onto the Softphone:

1. Click on the settings wheel next to “Enter Number”.

2. Navigate to “Select Account” and select the relevant SIP Account for your extension.

3. Once you have selected the account, the extension will register.

Register Failed on the Euphoria Advanced Softphone (Agent Manager)

• There is a working Internet Connection, but you cannot PING api.euphoria.co.za or tms.euphoria.co.za  or the PBX.

• The Agent Manager works with the TMS, if you cannot reach tms.euphoria.co.za or api.euphoria.co.za, your phone will not register.

Register Fail (503 DNS Timeout)

This means that your computer could not resolve the DNS through your Internet Connection.

Usually changing the DNS settings on your Router fixes this. Alternatively, if the customer’s PC is set to DHCP, changing it to a Static IP and DNS with the same IP Address, Gateway and Subnet Mask that the computer got via DHCP will work. Make sure that it is set to “Use the following DNS server addresses” and insert the settings as seen in the image below

After changing the DNS Settings on the PC, make sure to click “OK” to save the settings and then close and re-open the Agent Manager to see if it registers with the new settings.

Putty Into Cisco Router (Diginet Connection)

In order to putty into a Diginet router (Cisco) you need to open three Putty Sessions.

1st Session

1. Log into Vibe Headend - vibe6.euphoria.co.za port 22

2. Navigate to tunnels and add  8081 followed by 10.0.3.238:80 and 22 in the forwarded ports section. 

2nd Session

1. Log into primary connection - 127.0.0.1 8082

2. Navigate to tunnels and add 8083 followed by Diginet router (Cisco) IP (in this case it is 192.168.0.1) port 23 in the forwarded ports section. 

3rd  Session

1. Log into primary connection - 127.0.0.1 8083

Telnet

The overall result should look like this:

Vibe Headend entry highlighted in red

1. If it shows the Mac address, either the Vibe CPE is not connected correctly to the network, or the Firewall is blocking the VPN from exiting the network

2. If it shows the IP address, Packets are being sent to the VIBE server but are not able to return back into the local LAN. Please check the firewall, port forwarding and mirror.

Vibe router not up but showing that its up but its not up and phones are showing no service through vibe but not when you bypass it.

The head end might have cached information and when we remove the head end info and apply settings then re-add the info the link comes up.

Calls dropping using a Mikrotik OS. (Router)

We have seen a SIP helper cause occasional dropped calls when using a Mikrotik Router without ViBE.

1. Go to Firewall Settings

2. Go to Service Ports

3. Ensure sip on port 5060 and 5061 option is disabled.

4. When you mouse rolls over it, it will give you the option to enable, as it is disabled

Euphoria Telecom

Sonicwall Firewall blocking Inbound Dialling and Extension Status.

http://www.informaticapressapochista.com/asterisk/asterisk-with-sonicwall/

How to install and use Zoiper on WP ( Windows Phone)

Go to the app store and download Zoiper. After installation, open Zoiper and follow the below instructions.

1. Click on the '' I already have an account'' option.

2. Click on the ''Manual'' option.

3. Click on the ''SIP'' option.=

4. Enter the details as needed:

1. [You will find your domain on the TMS under extension manager - the extension you are setting up - ''Send SIP password'']
2. My SIP account = Your account name such as, '' Home'' for example
3. Domain = The PBX IP address you are on Eg: 41.221.5.246  (pbx5)
1. pbx.euphoria.co.za - (41.221.5.107)
2. pbx2.euphoria.co.za - (41.221.5.241)
3. pbx3.euphoria.co.za - (41.221.6.227)
4. pbx4.euphoria.co.za - (41.221.5.244)
5. pbx5.euphoria.co.za - (41.221.5.246)
6. pbx6.euphoria.co.za - (41.221.5.247)
7. pbx7.euphoria.co.za - (41.221.5.248)
4. Username: The extension's ''SIP'' username, Eg: 100-euphoria
5. Password: The extension's SIP password
6. Find the password by clicking on the ''Send Password'' button. ''Send Auth key'' (Will send an Auth key to the email account mentioned in the pop-up box for the SIP details) - Navigate to your email account and access your inbox -
7. Open the mail regarding the auth key - Copy auth key - Paste the auth key where required, then click on show password.
   

5. Insert the User name ( SIP username ) where it's asking for the ''Auth username.

6. Next step is to enable the G729 audio codecs, navigate to the settings wheel and click on premium features.

1. This is not compulsory and the softphone would work without the codec, the quality might just be a pain.
2. For this feature, you would have to purchase the codec.
   

7. Once you have paid for the codec you will now be able to enable that codec (and only that codec) and move it to the top of the list to make it the primary codec.

Issues with an Asterisk Installation with SIP Trunk behind a Sonicwall TZ100 Router.

The problem usually is the one-way communication router through one trunk or another related issue.To solve the issue there are the general rules:

1. Set the UDP timeout to 90 sec or more.
2. Do not use SIP transformations (Voip section) and modify the NAT behaviour.
3. Forward all the necessary ports to PBX in LAN.

How to Configure the Sonicwall TZ100.

7. Click on Voip, then Settings.

8. Check the flag “Enable Consistent NAT” e uncheck the flag “Enable SIP Transformations”.

The protocol used from Asterisk in SIP is UDP, that is connectionless, so the connection between the two ports (5060-14001) will be kept a certain time, because there is no way to know if the connection is terminated or not. For this reason the association will be maintained until a timeout: the default in Soniwall in 30s, less than the Asterisk default SIP registration refresh period of 60 seconds! We had increased this value more than the registration refresh period (90s).

Note: It is possible to change the Asterisk registration refresh period too, but I prefer this solution (change the configuration of Sonicwall).

All these changes are sufficient more often than not: for the unfortunate cases, then you need to directly forward all the ports used by the SIP flow communication directly from WAN to the PBX.

In the next we will redirect all the all the necessary ports to the PBX (5060/UDP and the range from 10000/UDP to 20000/UDP).

Firewall -> Service Objects

Create two new Custom Service Objects: PbxSipSegn & PbxSipStreamVoce.

Create one new Custom Address Objects using the LAN IP of the PBX (in my case 172.18.49.200).

After creating the necessary objects now let’s change the firewall rules: add a new rule WAN->LAN.

The last step: we have to create the NAT policy.

Externip = <External ip address>

localnet = <Network address of the LAN network/Subnet Mask>

In the trunk conf you must add the next parameter.

nat = yes

QoS Solutions

# To QoS for PBX's #

We accept UDP connections on port 5060 and TCP connections on port 5061.

You will need to have some kind of DSCP marking rule to tag voice packets for the destination of the PBX.

Pbx destinations are:

41.221.5.104/29

41.221.5.224/27

41.221.6.224/29

154.119.166.64/27

154.70.244.128/27

# To QoS for ViBE #

We accept UDP connections on port 65500. ( bidirectional )

You will need to have some kind of DSCP marking rule to tag voice packets for the destination of the tenant’s ViBE Server.

ie: {internal_lan} DSCP mark DST 41.221.5.229:65500 UDP route via alternate breakout.

# A note for softphones #

Should you be using the Euphoria Agent Manager/Softphone app, it makes use of our tms.euphoria.co.za and api.euphoria.co.za to authenticate. These destinations should be unblocked on your network. The TMS and API both have two destinations, the DNS works in round-robin fashion. It's wise to have both destinations for both services open as it could resolve to either.

The TMS and API operate on port 443 TCP, subnets are as follows:

41.79.37.32/28 

41.71.113.224/27 

41.71.119.0/27 

105.28.102.192/29 

105.29.154.0/27

Auto Provisioned Zoiper for Windows/Mac or Android/iOS

I don't know if it is ideal to send this link to clients, because it contains the username and password of the extension but here it goes...

To provision windows/mac use:

https://www.zoiper.com/en/page/a03f8b620171dbd212c32e50601da629?u={USERNAME}&h={HOST}&p={PASSWORD}&o=&t=&x=&a=&tr=

To provision Android/iOS (this will generate a QR code you can copy and paste in an email safely)

https://oem.zoiper.com/qr.php?provider_id=8917284811b74195bd3d7b19fb818487&u={USERNAME}&h={HOST}&p={PASSWORD}&o=&t=&x=&a=&tr=

Additional Section for Resellers?

Phone's not registering

1. Sometimes when a phone doesn’t register, it could be something simple as the SIP password that is incorrect (Recommended to copy/paste) or that the PBX server is not correct when setting up a phone. 

2. If all the details have been Copy & Pasted, you can start to look a bit deeper into the settings of the phone. Usually, you can start with registering the phone on Account 2.

3. Make sure the Network Cable and Network Port in which the phone connects to is active and gives out DHCP (unless network setup is only allowing Static IP Addresses, in which case you need to get an IP Address assigned to the device and set it up statically). 

4. Check that the cables are in working order as it could be that the network cable is Faulty.

5. Ensure that the network cable plugged into the phone is connected to the “Internet “port on the phone.

6. Are you using the correct Protocol? The Protocol should always be UDP 

7. Try changing the local SIP Port, found in Account Settings under the advanced tab or Under the Settings tab in the SIP category. The SIP Port will always be 5060 by default. You can change the SIP Port to something else between 5060 and 65500. 

8. Try moving the phone to a different network point where there is a working device. It could be that the network point is causing an issue.

9. Try and register the PBX domain directly on the IP address as there could be a DNS translation error. (Gateway router is not translating the PBX to the relative IP). This issue could be resolved by troubleshooting your network / gateway router.

10. If the above steps do not work, you could confirm if there are VLANs set in place on the customer’s network as well which the phones need to be on. 

11. In rare cases, the customer’s network can have issues as well, such as DNS issues. If you change the phones DNS to the following, the phone should register:

1. Primary: 8.8.8.8

2. Secondary: 8.8.4.4

If the phone does register after the changes have been made, the network would need to be looked into from the customers side.  

Completing the above steps and the phone doesn’t register, the customer would need to look into their network setup and check if there is any conflict on their end.

Disable IGMP Snooping.

Unable to call an extension:

1. If you call the extension and it goes directly to voicemail, it is probably on DND or offline. Disable DND and the call will go through. DND rejects ANY call sent to the phone. 
2. The agent could be paused In the Queue (Enterprise PBX only offers queues). However, this would only allow the extension not to receive external calls from the queue. 
3. Check with Euphoria Support if all relevant codecs were enabled for the extension.

Unable to call out but receiving calls

1. If it is an agent type extension, check if it is assigned to an outbound queue
2. Should you have a firewall on the network, it might be that the traffic is allowed to go out but not come back in. If you have implemented the firewall rules please ensure its bidirectional ( Allowed in and out )
3. Registration might have dropped from the network, sometimes rebooting the device might resolve the issue.
4. In some cases, The extension would be registered across multiple devices

ViBE Router not coming online

1. On some occasions, the ViBE will not come online because the password still remains “TenantName”. This password would need to be replaced with the PBX’s Tenant Name which can be retrieved from the TMS.

2. One of the obvious reasons would be that the Primary connection is not set correctly. 

3. By default, we usually set Port 2 to have a static IP 192.168.220.2 which connects to the primary internet connection (ADSL/LTE) which will be configured with a gateway of 192.168.220.1. If you have a dedicated internet connection or are going to share your internet between voice & data, you can plug that into port 3of the ViBE and either remove the route on the ViBE, or assign a reserved IP to port 3 of the ViBE and update the route. 

4. Make sure that you have purchased the license from the supplier and that the CPE is activated with this license

5. Check that the details sent to Euphoria are correct such as the Mac address and password.

6. Check if the ViBe config file loaded on the vibe is corresponding to the server where the Euphoria agent added it to  e.g. ViBE1, ViBE2, etc

You can refer to the Reseller Dropbox in order to find the steps on adding and removing routes on the ViBE Router. 

Call quality issues

1. This is mostly experienced when a customer is sharing internet with voice and data. In this case, you would need a dedicated Internet Connection for Voice. If you are using Fibre, it would be recommended to get a minimum of 2MB dedicated for voice (You would need to get in touch with your ISP and get them to do this on their end). QOS rules are provided on the Reseller Dropbox in the Network Configuration folder.

RSAWeb are able to implement a VLAN for Euphoria, you would need to contact them and ask them to set it up for you. 

2. On rare occasions, the ViBE can be set in Rain Mode on the CPE, but on the headend the ViBE will be set to Failover (Vice versa). You are welcome to call/mail Euphoria Support to confirm the server set-up. 

3. If you are experiencing call quality issues like crackling, a loud buzzing noise etc., try checking that the curly cord is plugged into the correct port.

1. Try calling on loudspeaker

2. Try changing curly cord

3. Try changing the hand piece.

4. Check firmware of device

5. Factory reset the phone 

4. If the caller says that they can’t hear you, experience one-way audio, that you sound like you are underwater or disappear etc., this is most probably a Firewall issue. You need to make sure that the correct protocols, sip ports and UDP ports are allowed through the firewall, as well as from the ISP side. (Ports and IP’s are provided on the Reseller Dropbox in the Network Configuration folder.

5. When there are call quality issues, you would need to confirm what kind of phone it is as well. Is it a desk phone/cordless phone/cell phone app/Softphone etc.? With this being said, if it is a cordless phone, make sure you are not out of range from the base or if there is anything else near it that can cause interference. 

If a ported number cannot be reached after it has been ported to us.

1. You would need to know if this is from a certain Network or from all networks, including phones on the Euphoria Network. 

2. If this is happening on all phones, you would need to make sure that the routing on the Euphoria’s side. Only if you get a message saying “I’m sorry, the number you have dialed cannot be found on the Euphoria Network”, it would mean that the number has been ported, but not been added to the Account yet.

3. If the routing Is setup on Euphoria’s network, get the latest example of a call that happened within at most a 6 hour time frame and escalate this to the relevant network provider where they will let you know what the issue is. On some occasions, the routing has not been updated on Vodacom, Cellc, MTN etc side (The relevant network provider  needs to sort this out on their side). Send all call examples to Euphoria Support so that they can investigate and escalate it to the relevant network provider if needed. 

Please do not contact the network provider directly, all correspondence with the network provider MUST be done through Euphoria Telecom.