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support
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Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: February 14 2008 at 3:31pm | IP Logged
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LanScape support received the following email from a customer. We wanted to post this information here in case it is useful to someone else. Thanks.
-----Original Message-----
From: xxxxxxxx@yyyyyyyy.com
Sent: Thursday, February 14, 2008 2:15 PM
To: zzzzzz@lanscapecorp.com
Subject: VoIP Dialer
From our application written in VB6, we need to call a PTSN phone using what ever methodes you have available. We also need help to determine the length of the call,whether the call was answered and wav file played. Please let me know.
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support
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Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: February 14 2008 at 3:36pm | IP Logged
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Hi Woody, :)
Thanks for your email. What you describe should not be a problem if we understand you correctly.
You >>>
From our application written in VB6...
<<< Support
The best way to go about this and add VOIP to a VB6 app is to use our VOIP Media Engine product:
http://www.lanscapecorp.com/ProductPages/LanScapeVoipMediaEn gine.asp
and "wrap" the needed APIs in a simple ActiveX COM object. We have already done similar things like this for other customers.
You >>>
we need to call a PTSN phone using what ever methods you have available.
<<< Support
If you need a very simple solution and you can terminate your own PSTN phone lines, you can set up your VOIP enabled VB6 app to talk to a single channel Linksys VOIP Gateway (Model # SPA3102). These go for under USD$100.00. You deploy your VB6 app on a host machine in your local network and configure your Linksys VOIP gateway to live somewhere else in your network. You VB6 app then makes PSTN calls (to other land line phones, cell phones, whatever) through the gateway. Simple.
You >>>
We also need help to determine the length of the call...
<<< Support
Not a problem.
You >>>
... and whether the call was answered
<<< Support
Not a problem.
You >>>
...and wav file played.
<<< Support
Hmmm.. not exactly sure what you mean here.
We are assuming that you want us to assist you on a contract basis. Is this correct?
We have taken the contents of this email and posted it to the support forum located here:
http://www.lanscapecorp.com/forum/forum_posts.asp?TID=438&PN =1&TPN=1
Please create a support forum user account so that we may continue this discussion there. You can remain anonymous.
Thanks,
Support
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Woody
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Joined: February 14 2008
Location: United States
Posts: 2
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| Posted: February 14 2008 at 4:31pm | IP Logged
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I'm not sure the hardware and dedicated phone line is going to work for us. The requirements for our customer should be
1) Our module/dll's installed
2) Have internet connectivity
We want to keep it really simple for them. Basically we want to loop through a list of phone numbers, call and play an audio file(PayYourBill.wav). :) If the user listens to the message we log it. If the user answers and immediately hangs up we log it. If the number is "no longer in use" we log it. I see VoIP SIP SDK packages out on the market requiring an IP Telephony Service Provider to set us up with an account to their SIP Proxy Server/PSTN Gateway. Do you provide the same? And if so, can we get a proof of concept?
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support
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| Posted: February 15 2008 at 7:06am | IP Logged
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Hi Woody,
Thanks for the additional information.
You >>>
Basically we want to loop through a list of phone numbers, call and play an audio file(PayYourBill.wav).
<<< Support
Not a problem.
You >>>
If the user listens to the message we log it.
<<< Support
No issue.
You >>>
If the user answers and immediately hangs up we log it.
<<< Support
No issue.
You >>>
If the number is "no longer in use" we log it.
<<< Support
Should not be a problem. However, this needs looking into regarding how the final selected PSTN service provider(s) supports this capability. We will assume the PSTN service provider you or your customer selects will support RFC2833 “line events”. See section 3.12 Line Events of the RFC. We also assume that your outgoing VOIP calls will use a lossy low bitrate codec (G729 for example). In this case, when your app makes a call and the “number is no longer in service”, we should receive the Special information tone (RFC2833 RTP payload 74) from the PSTN provider to indicate the connection could not be established. Here is the definition of the Special information tone” from the RFC:
Special information tone: The callee cannot be reached, but the reason is neither "busy" nor "congestion". This tone should be used before all call failure announcements, for the benefit of automatic equipment.
This RFC2833 RTP payload 74 can be handled by the VOIP media engine.
You >>>
I see VoIP SIP SDK packages out on the market requiring an IP Telephony Service Provider to set us up with an account to their SIP Proxy Server/PSTN Gateway. Do you provide the same?
<<< Support
No. You can select just about any PSTN termination service provider you wish. Google and you will find many.
You >>>
…And if so, can we get a proof of concept?
<<< Support
Sure. It would require your firm to contract with us for the development work.
Repost as needed.
Thanks Woody,
Support
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Woody
Intermediate
Joined: February 14 2008
Location: United States
Posts: 2
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| Posted: February 15 2008 at 9:03am | IP Logged
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Guys,
Thank you for the detailed and fast responses. How much development work are we talking about? Aren't my requirements just a small sample of the standard features offered on most any VoIP interface? Can you give me a ballpark figure to present to my team?
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support
Administrator
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Location: United States
Posts: 1666
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| Posted: February 15 2008 at 12:30pm | IP Logged
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Hi Woody,
You >>>
How much development work are we talking about?
<<< Support
The work would involve wrapping a small subset of the VOIP Media Engine API with a windowless ActiveX control you will be able to call from your legacy VB6 code. That’s the “clean” way to do it. Anything else would be a hack.
Also, develop a driver test app in VB6 that we would use to test all the functionality you require in your VOIP end product. When all the work is complete, your dev guys can then take the ActiveX control we will supply you in binary form and code it into your legacy app.
If your dev guys can develop the native Win32 C++ ActiveX wrapper, then your team should be able to do the whole thing by yourself.
You >>>
Aren't my requirements just a small sample of the standard features offered on most any VoIP interface?
<<< Support
Yes.
You >>>
Can you give me a ballpark figure to present to my team?
<<< Support
Not at this time. Other things to consider like: When do you want the work completed, what are ALL the requirements, things like that. If you think you would like us to assist in the development, then we can talk about all the details off line away from this public forum.
One thing that will be important to consider is: If you do not want to allow your customers to terminate their own PSTN line(s), then your firm will have to select appropriate PSTN carriers and we will have to test the VOIP functionality against their equipment to make sure of 100% interoperability.
Support
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Pre-Sales Technical Support
Topic: Using the Media Engine in a VB6 app to make PSTN calls
