GotSpeech.NET

Hi,

I am new to speech server 2007 and so far I ve managed to test my voice apps from xlite softphone.

I would like to access my voice apps (me and everyone else) from pstn network (normal phone). What are the hardware and sofware requirements to achieve something like that?

I ve heard about TDM cards. I am not sure what's their role. My guess is that they transform digital (sip) signals to analog (phone) signals. Am I right? 

Also I ve noticed that my softphone needs to be registered on the speech server in order to have access to my apps. Since I would like any phone number to access my voice apps then I guess that I need a proxy.

Do I need something more? Can someone provide some relative documentation or links?

Thanks in advance!
 

If you’re planning to do this in your home then you’re most likely looking for a basic analog to SIP gateway (either a card or an external appliance). If you’re in an office then you need to know what type of PBX you’re connecting too. Different PBX flavors have different connection options.

Assuming you’re looking for a home install that is drop-dead simple (and price isn’t a key concern) then I would suggest a Dialogic DMG1000. It is relatively easy to configure, well documented, and there is plenty of community support for it.

There are a number of less expensive ways to talk to your speech platform but they require a bit more research. Not all of those TDM cards convert PSTN to SIP; most use their own protocols and you need special drivers to get Speech Server to work with them. There are also 3^rd party SIP gateways like Paraxip which can add SIP support to Dialogic and Sangoma TDM cards.

What sort of environment are you looking to deploy in? I might be able to give better answers with some additional information.

 

Hello, Thank you for taking time to reply. First of all I would like to say that I am very new to speech server, voice applications or voip in general. Now, all I need to do is provide the possibility to the users to call my application (my single home phone number) from their wired phones (pstn) instead of just xlite (that I use for tests myself). My phone number will do nothing more than just serve the application (will not be used for answering calls). For that I believe a voip gateway like the one you mention above is needed. But I am studding the diagram on the webpage: http://msdn.microsoft.com/en-us/library/bb858792.aspx it indicates that for telephony clients an ip pbx is required. For voip clients a voip gateway is required… Can someone tell me if this is true? And if yes why is that? Let’s suggest that a voip gateway is all that I need to achieve that. What if I want to serve 2 or more (small number anyway) simultaneous calls? If 2 users call me at the same time one of them will get the busy tone. Am I right? Do I need extra software or hardware? Thank you again and sorry if my questions are too simple but I don’t know from where to start!

What you're talking about then is connecting a single analog phone line to Microsoft Speech Server. To do this you need a gateway that will turn your analog signal to SIP and vice versa. There are a number of products that you can use to do that.

One option is to use a Dialogic DMG1000 Analog. It has 4 analog phone jacks and one Ethernet port. You plug your phone line into one of the jacks, your LAN into the network port, and tell the gateway the IP address of the server you want to answer the call. On the Speech Server side you create a SIP Peer for the IP address of the gateway. That is all there is too it.

The part number for the gateway is DMG1004LSW. It goes for ~$900 (see http://www.google.com/products?q=DMG1004LSW&oe=UTF-8&scoring=r)

I have my home office setup the same ways as ml_ described.

To answer your other question - you won't have to trust your friends phones. You simply trust the gateway.

---

Marshall Harrison
Microsoft MVP, Office Communications Server


www.GoldSys.com
W: 303.447.2774
H: 904.342-6205

the gotspeech guy

thanks a lot ml_ and marshallharrison for replying!

one more question:

Lets suppose I have only one phone number and one line connected to the gateway you mention. Will I be able to serve two or three simultaneous calls or will the user get the busy tone?  

If not do I need extra hardware (and what), or extra phone numbers?

Thanx again and sorry for bothering again but I have no previous experience... 

It is always one call per phone line.

Something to start thinking about if you're going to build speech applications - phone lines and phone numbers are not the same thing. The number of phone lines controls the number of concurrent callers. Phone numbers are like DNS entries, you can have any number of them pointing at a single phone line.