There are a couple of notes about this however: It is best to have the IP subnet that the VoIP device on be on the same subnet as the DXL Exchanges.

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1 Our DXL system with the VoIP (Voice over IP) Option, in addition to the Harding VoIP masters, intercoms, and paging speakers, can interface to VoIP enabled devices such as PDA's and PC soft-phones. We use standard SIP session management and RTP voice protocols, using G.711 u-law codec. Pretty much every VoIP phone or soft-phone is going to have G.711 as it is a required uncompressed codec. There are a couple of notes about this however: It is best to have the IP subnet that the VoIP device on be on the same subnet as the DXL Exchanges. This also means that you cannot normally use a VoIP device over cellular data plans or internet even if the DXL systems do have a router and internet connection It is probably possible to go across subnets but that is something we currently cannot provide technical support for. Attached is a short write up on the settings you need on your VoIP phone or VoIP software to get it to communicate with the DXL system. The differences you see when using a standard VoIP phone or soft-phone rather than the DXL TMM-640 or IMS-640 master station are: The TMM-640 and IMS-640 are always connected in a voice path, while a standard VoIP phone is only connected when a call is connected. The TMM-640 and IMS-640 do this so that it can play "call queued up" beeps and other master status indication tones. This means: a. On the VOIP status screen of the DCC, TMM-640 and IMS-640 show as "RC" (Registered and Connected) all the time while standard VoIP masters show as "R" when configured but not in a call, and "RC" when in a call. b. A queued up call will beep on the IMS-640 or TMM-640 master (if configured to do so), while a queued up call on a third-party VoIP phone will ring the VoIP master with the VoIP phone's ring tone (whatever is defined in the VoIP phone) Here's some examples: Call request from station step 1: Station presses call request step 2: VoIP phone starts ringing, caller ID shows intercom number step 3: When you press the "answer" button on the VoIP phone, the connection is made and you can talk and listen to the station step 4: When you hang up the VoIP phone, the connection is disconnected

2 Host sends connection command step 1: Host sends "Ical" <VoIP_master> <station>" step 2: VoIP phone starts ringing, caller ID shows intercom number step 3: When you press the "answer" button on the VoIP phone, the connection is made and you can talk and listen to the station step 4: When you hang up the VoIP phone, the connection is disconnected VoIP phone dials intercom station step 1: VoIP phone dials "#<station>" then the "Call" button (I.E. "#101" then "Call" to dial intercom 101) step 2: The connection is made and you can talk and listen to the station step 3: When you hang up the VoIP phone, the connection is disconnected (dial keys are same as telephone masters - by default "#=intercom", "0=master", "*=page zone" plus others as configured in "System" settings) So: If you have a phone that you are not controlling with a host: - Answering calls is normal. Phone rings, you answer and start talking - Placing a call is done by dialing the call then the Call button. If you have a VoIP phone that is host controlled: - Call requests should go to host as a virtual master (not including this master, otherwise it will ring for call requests) - When host wants to answer call, send "Ical <master> <station>" as normal. - Phone rings, you answer and start talking. - You can disconnect from the phone or host. If you have a soft-phone that is computer or PDA-controlled you can have it do the answering: - Call requests should go to host as a virtual master (not including this master, otherwise it will ring for call requests) - When host wants to answer call, send "Ical <master> <station>" as normal. - If your software controls the VoIP program, The VoIP software will send your program an "incoming call" event. Whenever you see this, send it the command to "answer call". The API between you and the soft-phone is of course dependant on the VoIP stack you use. - Then you can talk with the VoIP software. - You can disconnect from the VoIP software API or the host. The G.711 u-law (uncompressed) codec we use takes about 85 kbit/sec upload and 85 kbit/sec download when in use (about 170 kbit/sec total) continuous on the network.

3 An example of using an iphone or ipod Touch follow. This example uses the Acrobits Softphone SIP program available from the App store. Some free apps such as Media5-fone can be used for testing but usually have limitations such as ending a call after one minute. First use the DXL Administrator and create an IP master. Set Type to IP Master and pick the exchange (DCC) that it will connect to. Remember the IP address of that Exchange, you will need it in the configuration later.

4 On the iphone or ipod Touch, set up and use a wireless connection that is on the same subnet as the DXL system. This might be by setting up your wireless router for DHCP on that network, or setting up your iphone or ipod touch with a static IP address. To set up the SIP phone, start up the SIP program (Acrobits Softphone) and create a new account of type Generic SIP Account. Enter the account details: Title: Enter a descriptive name for the account (this is cosmetic and optional) Username: Enter the master number (this is required) Password: leave blank (or enter master number if you cannot leave it blank) Domain: Enter the DXL DCC s IP address (this is required) Display Name: Enter a descriptive name for this master (this is cosmetic and optional) Make sure that the account can be used for outgoing and incoming calls (no changes are necessary)

5 Exit to the main phone menus. The SIP phone should register then indicate it is connected (icon on top left). If it is not then you should check the master number in Username and IP address in Domain, and that the iphone or ipod Touch is on the same subnet as the DXL system. Now you can use the SIP phone program. A call request will ring the SIP phone and show the intercom calling in and you can answer the call. Depending on the program you may see different parts of the Caller ID. In this case this soft phone shows the name part of the caller ID only, some other phones also show the intercom number (which would be #101) I Station 101 is the caller ID I = Intercom Station 101 = name of station You can dial up on the main keypad screen by dialing the number then the call button. #xxx dials intercom xxx *xxx dials page zone xxx 0xxx dials master xxx