Come utilizzare il servizio di audioconferenza

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1 Come utilizzare il servizio di audioconferenza Il sistema di audioconferenza puo essere utilizzato in due modalita : 1) Collegamento singolo. Si utilizza l apparecchio per audioconferenza che si ha a disposizione come un normale telefono vivavoce, chiamando direttamente il numero della persona con sui si deve avere la conversazione telefonica oppure facendosi chiamare da questa. E bene tener presente che in questa modalita potra connettersi all audioconferenza un solo utente e che il costo della chiamata sara uguale a quello di una normale telefonata (urbana o internazionale). 2) Audioconferenza. E' possibile richiedere al CNAF una conference room inviando una a (preferibile) o chiamando telefonicamente le seguente persone: Stefano Zani Alessadro Italiano Marco Bencivenni Potranno essere assegnate due tipi di conference room, una temporanea (per audioconferenze occasionali) e una permanente (per audioconferenze periodiche). E possibile collegarsi ad una audioconferenza in due modalita : 1) Utilizzando un normale telefono e chiamando il numero Con questa modalita il costo della chiamata sara pari ad una telefonata urbana. 2) Utilizzando un cliente software/hardware e configurando in pochi passi un sip/iax client. In questa modalita la chiamata sara gratuita. Le istruzioni per la configurazione di un sip/iax client sono disponibili al seguente link:

2 Come configurare un sip client

3 A soft phone is an IP telephone in software. It can be installed on a personal computer and function as an IP phone. Soft phones require appropriate audio hardware to be present on the personal computer they run. This can either be a sound card with speakers or earphones and a microphone, or, alternatively a USB phone set. Soft phones are inferior to hard phones but cheaper to obtain, many are available as a free download. Personally, I suggest to use: X-Lite for Window Sjphone for Linux An Ethernet hard phone is a self contained IP telephone that looks just like a conventional phone but instead of a conventional phone jack, it has an Ethernet port through which it communicates directly with a VoIP server, VoIP gateway or another VoIP phone. Since a broadband hard phone communicates directly with a VoIP server, VoIP gateway or another VoIP phone it does not require any personal computer nor any software running on a personal computer to make or receive VoIP phone calls. It can be used independently, all that is required is an internet connection. While PC based software solutions are cheaper, a hard phone is the best solution for IP telephony. A low impact solution is rapresentad by Analog Telephone Adapters, better noted as ATA. Read here Market offers also: Cordless Hard Phones with IP interface on their base station. WLAN or WiFi Phones are hard phones with a built-in WiFi transceiver unit instead of an Ethernet port to connect to a WiFi base station and from there to a remote VoIP server Other info here Please note: you are free to use any SIP compliant client, just use the following parameters:

4 SIP proxy phone.infn.it port 5060 Login: 2004 Password: 5555 Here you can find some useful numbers Here some NAT/Firewall infos Here if you have audio problems: background noise, echo, etc Alfredo Pagano 14/03/2007

5 X Lite for Windows

6 Download and install it Add an account (as Domain use: phone.infn.it) Insert the correct parameters (see the image) click Ok and enable it (Check Enabled ). Test the environment calling *43 for the echo test. And remember to MUTE yourself when you are not speaking. Here you can find some useful numbers. Here some NAT/Firewall infos. Here if you have audio problems: background noise, echo, etc Here for the complete error code list

7 You can find a complete User Guide here (pdf file) Alfredo Pagano 14/03/2007

8 Sjphone for Windows and Linux

9 Step to step guide to install and configure SJPhone on Windows and/or Linux: STEP 1 Download the software from STEP 2 After installing, the default soft phone will appear. For Windows version: Click on the screwdriver symbol For Linux version: Click on "Phone -> Preferences" Tab The *Options* window will appear. Enter your name and address in the 'User Information' Tab. STEP 3 Click on the 'Profiles' Tab. In the Profile Tab click on 'New'. When the 'Create New Profile' window appears, in the 'Profile name:' enter INFN and in the 'Profile' choose 'Calls through SIP Proxy' Then click OK.

10 STEP 4 A 'Profile options' windows will appear. Click on the SIP Proxy Tab. Enter in 'Proxy Domain:' phone.infn.it and enter 5060 in the field to the right side of it. Under 'User domain:' enter phone.infn.it. In the same SIP Proxy Tab's Advanced options, put a checkmark on Use separate register. Under this enter in 'Register domain:' phone.infn.it and enter 5060 in the field to the right side of it. Click OK.

11 STEP 5 Back in the 'Profiles' Tab you will find an varphonex.com profile. Highlight INFN and then click on 'Initialize.." A 'Service:INFN' box will appear. Enter in 'Account:' 2004 and in 'Password:' 5555 and click OK. STEP 6 In the 'Profiles' Tab again highlight INFN profile and then click on 'Use...'

12 STEP 7 Click on the 'Call Options' tab. Make sure that under 'Incoming Calls' put a check mark on 'Automatically accept incoming calls'. In 'Outgoing calls' put a checkmark on 'Use following host address' and make sure you choose the IP address of your computer. STEP 8 Click on 'Audio' Tab and make sure that in 'Sound devices' the Playback: is using your sound device. Click OK. STEP 9 Call your conference number!! And remember to MUTE yourself when you are not speaking.

13 Here you can find some useful numbers. Here some NAT/Firewall infos. Here if you have audio problems: background noise, echo, etc Alfredo Pagano 14/03/2007

14 Come configurare un iax client

15 IAX is the Inter-Asterisk exchange protocol used by Asterisk, an open source PBX server from Digium. It is used to enable VoIP connections between Asterisk servers, and between servers and clients that also use the IAX protocol. If you want understand more, start to read here In few words, IAX2 is better than SIP because: IAX2 uses just one UDP port, 4569 (this solve most of the NAT problems) IAX2 supports PKI-style authentication and trunking. IAX2 uses only 2.4k for a single call I recommend: IDEFisk: A Windows and Linux softphone from asteriskguru.com Click to enlarge Download and lunch it. Configure it following this screenshot 1. Just lunched

16 2. Open the Option Menu by Right Clicking on Idefisk 3. Select "Account Options" and fill the camp. Click to enlarge Username 2003 Password 5555 And remember to MUTE yourself when you are not speaking. (Click on Microphone symbol that will become red, click again to Unmute) You can find a complete function list here (pdf file)

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