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Configuration passerelle VOIP Groupe LELAS

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CONFIGURATION GUIDE

VOIP SIP MODULE HT801

MT302SIP-POE

Groupe LE LAS COMMUNICATING IN SAFETY




CONTENTS I GENERAL PRESENTATION .............................................................................................3 II HARDWARE INSTALLATION ........................................................................................4

1- Telephone Initialization ..................................................................................................5 2- VOIP Configuration of gateway .....................................................................................7

III CONFIGURATION OF SIP SERVER ............................................................................ 13

1- Installation of server ..................................................................................................... 13 2- Between two VOIP units .............................................................................................. 17




I General Presentation The VoIP telephone can establish a communication telephonic through an Ethernet LAN via a SIP server THE VoIP telephone TLA402 series is equipment with a gateway VoIP module, which support to make a convert VOIP protocol to an analog line standard telephone. The VOIP phone is equipped with a phone card to transfer to the voice gateway and the frequency DTMF. Then the gateway converts the analog information into digital information to be sent to the SIP server in order to deliver these information packets to another IP phone on the network LAN. The phone be called receives the packets TCP / IP by its gateway, it will then convert the digital information into analog information to restore the voice in the telephone’s loud speaker.

Figure 1: Flowchart of the system. Example : The phone N°1 wants to call the phone N°10, then the user enters the number 10 on the keypad of the phone (1) the SIP server can establish this connection with the phone 10. The digital packets routed (from 1 to 10) contains the digitized voice, data is sent to the phone 10 for converting analog data.

VOIP TELEPHONE

(10)

SERVER SIP

LOCAL RESEAU

ETHERNET

VOIP TELEPHONE

(1)

Connexion Ethernet

Connexion Ethernet

Connexion Ethernet

Routing packets




Figure 2: Flowchart of VOIP phone.

II Hardware Installation Before connecting the VOIP phone on the Ethernet LAN, it must first configure an address range dynamically (DHCP) router on the LAN, because the VOIP phones are by default configured to use DHCP, It is to say that they take one of the IP address in the DHCP range. Then connect the network cable on the RJ45 connector of the bridge and the AC power cord.

Photo 1 : SIP/VOIP/PoE card - Réf.: MT302SIP-POE.

Telephone card

VOIP Gateway

Loud speaker card +

microphone

Push Button Carte

Voice

Push button Informations

Analog information

Digital Information

INPUT NETWORK

TELEPHONE CABLE




1- Initialization of telephone By default, the VOIP phone is dynamic address. In order to know the IP address of the telephone,there several solutions:

Ø Solution 1: If the telephone has a digital keypad, so just pressing the button to take the phone line, then press 3 times the button *. The speaker then broadcasts "Enter a menu option" then dial 02, the loudspeaker broadcasts the IP address of the phone.

Ø Solution 2: If the telephone has no keypad, disconnect the phone cord and

plug to an analog phone with keypad. Then take the line with the phone and press 3 times the button *. The speaker then broadcasts "Enter a menu option" then dial 02, the loudspeaker broadcasts the IP address of the phone. Then reconnect the phone cable to the original post VOIP. To enter a fixed IP address must follow the following procedure :

Take off the handset of telephone then press 3 times *

Menu

Announcement of voice options

Principal Menu « enter a menu option » Press * for menu following Press # previous

01

« DHCP mode » « Static IP mode »

Press 9 for change If using Static mode configure the IP address in menu 02 and 05.

02

« IP adresse » + IP adresse

The IP address is announced. If "Static IP address used" enter the 12 digit new IP address

03 « Subnet Mask » +IP adresse Idem except 02

04 « Gateway » + adresse IP Idem except 02




Once the IP address of phone known, it must enter into a PC web browser the IP address of the phone so that connect to a web server on the phone . The default password is "admin".




2- Configuration of gateway VOIP Click on the tab STATUS :

MAC ADDRESS : MAC addressee of gateway IP IP ADDRESS : IP addressee of gateway REGISTERED : YES if validated by the SIP server




Click on the tab ADVANCED SETTING :

Admin Password : to change the password for gateway configuration SIP Server : enter the IP address of SIP server on the LAN Ethernet. Outbound Proxy : enter the IP address of SIP server. SIP User ID : enter the phone number (eg 105) Authenticate ID : identification number from the phone enter the same number as before. Authenticate Password : password credentials to the SIP server. (ex: admin) Name: name of the phone (your choice). Preferred Vocoder : audio compression standard G723 rate : voice compression rate




iLBC frame size : size of the buffer the audio codec, default 20mS. iLBC payload type : Keep 97. Silence Suppression : it can send packets during a silent voice when compression G723 or G729 is selected. Voice Frames per TX : size of a packet voice network. Set 2 by default. Fax Mode : By default T.38 (Auto Detect). Layer 3 QoS : 48 by default. Layer 2 QoS : 0 and 0 by default. Allow incoming SIP message: select YES Use DNS SRV: NO User ID is phone number : allows a number of affiliate PSTN by default NO SIP Registration : YES,The telephone sends the information to the SIP server to make it through it is running. Unregister On Reboot : NO. Register Expiration : refresh the status information for connecting to the SIP server in seconds, default is 3600 seconds. Early Dial : NO. Allow outgoing call without : NO.




Use # as Dial Key : # key is used to validate the number dialed on the keyboard. Default Yes. Local SIP port : port of communication, defaut : 5060. Local RTP port : 5004. Offhook Auto-Dial : Provide call a number by the hook off.




Automatic Upgrade : When changing Firmware to Yes and enter 1 minute so you can download the new Firmware. The download is automatic. After a few minutes (5min) the firmware is updated, then it's back in the configurations of the bridge and check NO and enter 10080 minutes, and never update the firmware.




Above configurations allow users to adjust various parameters of tone telephonic. Put those parameters setting in your gateway. After setting the gateway following the instructions above, it must proceed to adjust the SIP server. We recommend using a SIP server downloadable from 3CX (free) at: http://www.3cx.fr/pabx/free-edition.html , The Software : 3cxphonesystem2.exe .




III Configuration of SIP server A Software PBX-IP provide many advantages:

Ø No separate telephone connection required - telephones use computer network Ø Management and installation easily via the web interface configuration Ø An Software IP PBX is less expensive than hardware PBX Ø Employees can move offices without worrying about connections or

configuring the IP PBX

1- Installation of server

Double click on the icon for installing the software

Then click NEXT, then follow the instructions specify a username and choose a password to return to the SIP server configurations.




Once the server installed the following page is displayed:

Enter the username and password previously returned when installing the software.

At first right click on the tab "Status Line". We can see no telephone VOIP there is referenced. We must therefore add. Click below the tab on "ADD".




Enter a number in Extention number, eg 105 (this is the number that was registered in the gateway configuration in II-2), then enter a name choice. In the authentication settings enter the number 105 (serial gateway) and his password (previously stored at the gateway in II-2, "admin"). Let the other in defaults and click OK

This window opens, indicating that the post 105 is created.




Then click the tab Line Status

The telephone N°105 is functional because their logo is in green, if the logo is red it means that the position is not recognized, this could be due to a problem "password Authenticate" between the gateway and the SIP router, make sure the two passwords are identical. Otherwise do a reset of the gateway via the web page server. Once you record a new telephone, you must to reset the telephone. To create new telephone to the same manipulation as before. Example : two telephones actives at LAN network:

In the top right of the Line Status can see the status of the SIP server. : ONLINE : operation. OFFLINE : No operation




2- Between two VOIP units Telephones

To perform this test requires that the SIP server software is ONLINE, it is also that IP phones are detected by the server (Phone xxxx = registered).

Photo2 : Connection Telephone VoIP on the LAN Ethernet

The telephone with keypad has been set up as number 105 and the telephone with 1-button has been configured as number 106. Option offhook Auto-Dial of telephone 106 was set to 105, which is telephone number with keyboard. Thus, by press of button of telephone106, the phone call ext 105. To call the phone with 1 button , you must enter the keyboard combination ext 105 # 106.

Phone VOIP 1 Button

Phone VOIP with keypad.




GROUPE LE LAS - PARIS

FRANCE 34/36 RUE ROGER SALENGRO F 94134 FONTENAY SOUS BOIS

Tel : 33 01 48 76 62 62 Fax : 33 01 48 76 83 04

Internet : www.lelas.fr E-mail : [email protected]

Edition C 07.03.2019

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