Configure a SIP Trunk
1. Create an Account with the SIP trunk provider.
To start with, you need to have an account from your SIP trunk provider.
Yeastar specialise in the design and development of innovative telecommunications equipment, including VoIP PBX systems and VoIP gateways for the SMB.
Download the configuration templates for the following Yeastar SIP compatible IP-PBX
Provider Type | Supported Codec | Supported DTMF | Supported Fax Type | Link to Support Document | Tested Version |
Register Based | a-law, GSM | RFC2833, Info, Inband, Auto | T.38 | 30.6.0.16 |
1. Create an Account with the SIP trunk provider.
To start with, you need to have an account from your SIP trunk provider.
2. Add a SIP Trunk in P-Series PBX System
After you get the SIP trunk account, you need to add a SIP trunk in Yeastar P-Series PBX System.
Go to Extension and Trunk > Trunk, click Add.
3. Configure the trunk
Basic
Detailed Configuration
The parameters of the certified ITSP template are embedded. You don’t have to figure out Trunk Type, Transport, Hostname, Port, Domain (some ITSP partners’ server is not static,then you have to fulfill in yourself with the last server address).
4. Check the Trunk Status
Click Save and Apply. Check if the trunk is connected in Status.
To specify how calls from the SIP trunk should be routed, you need to configure an inbound route for the SIP trunk.
1. Create an Inbound Route
Go to Call Control > Inbound Route, click Add.
2. Configure the Inbound Route
3. Click Save and Apply
When you call in the SIP trunk, the call will be routed to the destination configured on the inbound route.
To make outbound calls via the newly created SIP trunk, you need to configure an outbound route for the trunk.
1. Create an Outbound Route
Go to Call Control > Outbound Route, click Add.
2. Configure the Outbound Route
The system compares the number with the pattern that you have defined in your route 1. If it matches, it will initiate the call using the selected trunks. If it does not, it will compare the number with the pattern you have defined in route 2 and so on. The outbound route which is in a higher position will be matched firstly.
You can adjust the outbound route sequence by clicking these buttons
3. Click Save and Apply
Now you can make outbound calls through the SIP trunk. As the dial patterns configured above, you need to dial “8” before the destination number.
For example, to call the number “01234567”, you need to dial “801234567” on your phone.
Gigaset Communications have created a process that enables Tesco Internet Phone subscribers with Siemens Dual-Phone DP450 to unlock the phone from the Tesco Internet Phone service. This will enable the phone to be used with VoiceHost.
This is an approved method for unlocking the Siemens Dual-Phone DP450 defined and released by Gigaset Communications, the original supplier of the phone to Tesco.
If this procedure is not followed exactly your phone might not work any more. In this case, Gigaset Communications or VoiceHost cannot accept any warranty claims.
NOTE: The Tesco Internet IPA 1000 ATA was an IAX2 device, most VoIP providers are SIP only, including VoiceHost.
Unlock procedure
Your Phone should have the latest released Tesco software version, else this unlock procedure will not work.
On your handset:
BASE SETTINGS SOFTWARE UPDATE [OK]
→→
Select and press OK Enter the system PIN (0000) and press [OK]
The base station establishes a connection to the internet or a local PC.
[YES] Press the display key to start the firmware update. If no update is needed, this will be displayed.
To unlock the WEB interface of the DP450, you need to enter the following service code. Please use the exact code as is described below, else your phone will not work any more.
SETTINGS BASE
→
94762001
(Now "Eeprom" must be displayed)
Insert the following number sequence:
01778 06183 06191
Press OK to finish the programming.
"Saved" must be displayed on the Handset.
Then release the phone to be able to upgrade it with the latest Retail firmware version.
SETTING BASE
→
94762001
(Now "Eeprom" must be displayed)
Now insert this number sequence:
04850 04351 04864
Press [OK] to finish the programming. "Saved" must be displayed on the Handset.
After this procedure, you can upgrade your phone to use the latest retail firmware version.
SETTINGS BASE SOFTWARE UPGRADE
→→
Select and press Enter the system PIN (0000) and press [OK]
The base station establishes a connection to the internet or a local PC.
[YES] Press the display key to start the firmware update.
Please Note:
The firmware update can last up to 3 minutes.
Some people experience problems with the firmware upgrade where the process does not complete. If after 30 minutes your handset keeps flashing 'Base 1' you will need to power cycle your phone base by disconnecting the power for 10 minutes and then reconnect it.
Allow the phone to boot, we recommend leaving it a further 30 minutes to boot and upgrade. If your handset is still flashing 'Base 1' after this time, you will need to pair the handset to the base. To do this, follow the procedure below:
On your handset:
SETTINGS HANDSET REGISTER
→→
On your base:
Hold down the blue button for 30 seconds.
If all goes well, your handset should now be paired and you should be able to make a call.
Once your phone has been unlocked, you will be able to follow our standard set-up guide:
Siemens Gigaset - Wizard
The Siemens Gigaset phone can be configured to use the VoiceHost service in two ways, the first method is by using the built-in configuration wizard on the phone handset. When you connect your phone to your router and power for the first time the wizard will automatically start. To configure the phone, you will need your extension username and password. Please note, these are not the details you have to log in to the website. The required details can be located in the control panel under 'Extension Config'.
Follow the on-screen prompts, select VoiceHost and enter your extension username and password. Once you have finished answering the questions in the wizard you should be able to start making calls with VoiceHost. To test your configuration, try dialling 1234 to listen to the welcome message.
If you require more than one account/extension configured on your phone, or you are having problems accessing voicemail or IVR menus you will need to check your settings by running through the following procedures.
Web-Based Configuration
To access the web-based setup, you will need to determine the IP address of the phone. To do this you will need the handset. On the handset select 'menu > settings > base > network > IP address. The display will show you the current IP address of the phone.
Now you have obtained the IP address you can enter this address in a web browser to access the web configuration page.
Enter the default system pin of '0000' and click ok. Once logged in you will be shown a menu. Select the 'Settings' option and then 'Connections'. Select the first SIP Account 'Edit' button. You will be shown the following screen:
Please enter the following details into the fields provided:
- SIP:
Authentication Name: <extension username> See Control Panel for Details
Authentication Password: <extension password> See Control Panel for Details
Username: <extension username> See Control Panel for Details
Domain: See Control Panel for Details
Display Name: <your name>
Proxy server address: See Control Panel for Details
Proxy server port: See Control Panel for Details
Registrar server: See Control Panel for Details
Registrar server port: See Control Panel for Details
- Network:
STUN enabled: no
STUN server: See Control Panel for Details
STUN port: 5000
NAT refresh time: 20 seconds
Outbound proxy mode: Auto
Outbound proxy port: 5060
- Voice Codecs:
VoIP volume: Normal
Enable Annex B for G729: No
Selected codecs: G711 ulaw and alaw
Once the above details have been entered select the 'Set' button to save your settings.
To make sure that your 'touch tones' or DTMF will work correctly, select the 'Advanced Settings'. You will be shown the following screen:
Change the 'DTMF over VoIP Connections' to 'SIP Info' with all other options unticked. Click 'Set' to complete the setup process.
Once the phone registers, you will be able to start making calls.
(handsets are simply paired to the base station)
Power up the N300IP, N510 or GO DECT Base station and connect a network cable into the side. Once the connection light comes on a steady blue then the device is powered up, connected to the network, and ready to use.
The base stations can have different Gigaset Model handsets paired to them, therefore, the handset model does not matter.
IP DECT Handsets for use with Gigaset Base Stations are:
If you need to set the base station with a static IP then set the IP Network details here, if it is OK to have it as DHCP then leave the settings on this page as they are. Please note that if you change to a static IP address then you will need to reload the web page.
Your Base Station and handsets should now be configured and working.
Below Version 8: Avaya IP Office software older than version 8 requires a STUN server for systems that do registration.
Navigate to System in the tree-view menu on the left of the PBX GUI
LAN Settings - Enter either the public IPv4 address of the broadband circuit or one that has been issued to you the customer to use. (STUN server details can be entered here)
IP Route - An IP Route will need to be created containing the IPv4 Address of the VoiceHost Proxy, giving the route to access the gateway connected to the LAN 2 port.
SIP registrar - These settings are normally left as default
SIP Line:
SIP URI settings for outgoing calls
SIP URI Setting for Incoming DDI Calls.
Incoming Call Route
A Pcap or 'Packet Capture' is also known as a log or SIP trace. As the name suggests it simply records all of the communications data and allows for it to be loaded into applications for analysis. Along with an understanding of SIP it is possible to ascertain where faults and errors occur. The location of the Pcap is important as VoiceHost can obtain them from our edge network but they can also be of value locally (from inside a local network or behind a firewall).
TLS and SRTP traffic will require decrypting and for obvious reasons we do not share encryption keys.
PCAPS also fall under various data protection laws legislation including GDPR and RIPA. You must adhere to legislation as do we.
Two programs are useful for this purpose but bear in mind that TLS and SRTP traffic will require decrypting and for obvious reasons we do not share encryption keys.
Collectors:
Analysing:
Let's describe the most important message headers in the example above.
The request start line: The string "INVITE sip:13@10.10.1.13 SIP/2.0" tells that this is an invitation to a call. It also gives the SIP address of the receiving endpoint (sip:13@10.10.1.13) and identifies the version of the protocol (SIP/2.0).
The message body carries a message of the Session Description Protocol. This message contains a description of the audio (and possibly video) channel that the calling endpoint wants to establish. We will look at the Session Description protocol later in a greater detail.
SIP Requests
SIP originally only had 6 requests (also called methods). These requests have been a part of the standard since SIP 1.0.
Individual methods can be described as:
In addition to these six requests, several other SIP methods have been added, either in SIP 2.0 or in other individual RFCs. For example, the INFO method was defined in RFC2976. It can be used to carry application-level information that is relevant to the session, for example, participant images or account balance information. The INFO method can also be used to carry DTMF digits.
The three methods SUBSCRIBE, NOTIFY, and MESSAGE extends SIP with instant messaging features. If you send the SUBSCRIBE method, you are asking the other party to send you notifications about status changes ("available", "busy", "away", etc.). The status change notifications are then sent in the NOTIFY messages. Last, the method MESSAGE is used to send instant messages. The text of the instant message is simply transported in the body of the method (Content-Type: is usually text/plain).
OfficeServ Configuration
All other entries can be left in default
DDI Table –MMC714/DM3.2.3
Use the full inbound number but replace the 0 with 44
VoIP Options – MMC501/DM5.2.18
To properly support early audio on outgoing calls, e.g. BT Caller Redirect numbers (<NU TONE> “the number
called has been changed to...”), change the following setting:
Real Ringback - ON
Telephone calls can be dropped for a variety of reasons but if you're suffering regularly then there are few things you can check and remedy to resolve them.
SIP session timers tell the SIP Servers that the calls is still alive as looking for a BYE message is not reliable. If a SIP Server thinks one end has hung up (i.e stops receiving KEEP-ALIVE messages) it will drop any open call legs.
Read our guidance on SIP ALG HERE
Talk Off happens when phone tones (DTMF) are wrongly interpreted during a call. This is easily resolved by ensuring the recommended settings for DTMF transmission OUT-OF-BAND as per RFC2833 are enabled.
If in doubt, turn this off as it essentially saves bandwidth by not sending full audio if nobody is talking but not everyone can support it.
SIP doesn't like NAT. It's that simple. Most people have to use it as IPv4 limitations dictate and few providers fully support IPv6.
BLF or Busy Lamp Fields are typically a collection of indicators on a phone that show who is talking on other phones connected to the same PBX and typically used by a receptionist or secretary to aid in routing incoming calls.
To configure phone keys via the Hosted Platform simply do the following:
You can download a formatted Key Labelling template for your Telephone Handsets below:
NOTE: These are pre-formatted to replace the paper BLF name inserts on your phone and should not be altered in size or scale.
Manufacturer | Model (click to download .doc .pdf) | Model | Model |
Snom | 3xx Series | 7xx Series | 720 | D725 | 760 | D765 |
Cisco | SPA500S Attendant Console sidecars | 7861 | |
Polycom | VVX Expansion Module | ||
Yealink | SIP-T38G , SIP-T28P , SIP-T26P , IP Phone Expansion Module EXP38 | ||
Grandstream | GXP 1628 | 1630 | ||
Panasonic | KX-UT133 | KX-UT136 |
Under 'Quick Dial' add the contacts you wish to monitor using the method prescribed at the top of this page.
Detailed Softphone instructions can be found here: https://www.voicehost.co.uk/help/softphone
BLF or Busy Lamp Fields are typically a collection of indicators on a phone that show who is talking on other phones connected to the same PBX and typically used by a receptionist or secretary to aid in routing incoming calls.
To configure phone keys via the Hosted Platform simply do the following:
Tentative Version 0.1(PSN) 18th, July, 2013
SIP Trunk – Port Property:
Important Note: Programming the details of the SIP trunk is done in this field.
In this example, the system has been programmed to use the changed FAX setting and NAT Keep Alive ability.
- Reject T.38 Request change to “Enable”. (Default: Disable)
*Note SIP server does not support T.38. (Need to set reject T.38 request by PBX.)
Recommended setting
- NAT - Keep Alive Packet Sending Ability change to “Enable”. (Default: Disable)
Go to 1.Configuration - 1.Slot and select “IPCMPR Virtual Slot”. and click “Ous”.
Move mouse over “V-SIPGW16” and click “Port Property”.
Main Tab:
1. Channel Attribute: Basic Channel
2. Provider Name: Enter a logical name
3. SIP Server Location – Name: Enter your assigned server as shown in the VoiceHost control panel.
4. SIP Server Location – IP Address: Not required
5. SIP Server port Number: Leave at 5060
6. SIP Service Domain: Not required
7. Subscriber Number: Not required
Account Tab:
1. User name:
Enter the SIP Account (User name) as supplied by VoiceHost. Please note that this is just the SIP Account (user name) and DOES NOT include @FQDN For example: SIP Account (User name) = ST00000T000 Enter: ST00000T000
2. Authentication ID:
Enter the Authentication ID as supplied by VoiceHost. Please note that this is just the Authentication ID and DOES NOT include @FQDN For example: Authentication ID = ST00000T000 Enter: ST00000T000
3. Authentication Password:
Enter the Password as supplied by VoiceHost
Register Tab:
1. Register Ability: Leave at Enable
2. Register Interval: Leave at 3600
3. Un-Register Ability: Leave enabled
4. Registrar Server – Name: Not required * If SIP Server and Registrar Server are different, enter the Registrar Server.
5. Registrar Server – IP Address: Not required
6. Registrar Server port number: Leave at 5060
Go Back to “Slot”.
Move mouse over “V-SIPGW16” again, and click “Shelf Property”.
NAT - Keep Alive Packet Sending Ability: Change to Enable
NAT - Keep Alive Packet Type: Confirm Blank UDP
NAT – Keep Alive Packet Sending Interval: Confirm 20
Then, click“OK”. Move mouse over“V-SIPGW16” again, and click “Ins”.
Incoming Call Routing:
Go to “10. CO & Incoming call” and select “3.DDI /DID Table”
1. DDI/DID Number: Enter the DDI number in the format 44+PSTN Number (as below)
2. DDI/DID Name: Determined by the installer (optional setting)
3. DDI/DID Destination: Determined by the installer (extension number, group etc)
All other settings can be left at default
Outbound Call CLI:
Each extension that wishes to present an individual CLI needs to be programmed with a usable CLI. The usable CLI is a PSTN number assigned with the SIP trunk.
i.e. if the PSTN number is 0843-9999999, the CLI to be programmed is 08439999999
Go to “Calling Party” tab.
1. From Header – User Part: Change to PBX-CLIP
All other tabs may be left at default:
- Header Type
- From Header – SIP-URI (100 characters)
- P-Preferred-Identity Header – User Part
- P-Preferred-Identity Header – SIP-URI (100 characters)
- Number Format
- Remove Digit
- Additional Dial
- Anonymous format in “From” header
- P-Asserted-Identity header
Go to “4.Extension, 1.Wired Extension, 1.Extension Settings” & select “ISDN CLIP”
1. Enter a valid CLI for each extension that requires it in the CLIP ID field. This setting, callee side shown ‘08439999999’.
2. Enter the name for each extension that requires it in the Extension Name field
This setting, what characters shown callee side is now testing.
[T.38 Tab]
1. Reject T.38 Request from Network: Change to Enable
All other tabs may be left at default
- T38 FAX Max Datagram
- T38 FAX UDPTL Error Correction - Redundancy
- T38 FAX UDPTL Redundancy count for T.30 messages
- T38 FAX UDPTL Redundancy count for data