vicidial.org

hi,
dont know whom to tell this, so posting here. Look at this http://www.packet-shaper.com/product/pd.html and http://voip-forum.tmcnet.com/voip-forum ... p?TID=8076 . These guys are selling this as an "indigenously designed and developed" product. However if you look at the flash demos of the admin/agent screens, there are a lot of similarities with the vicidial, infact in one of the screens (the "user status" screen in admin interface) you can see vicidial admin written there. They have tried to modify the look and feel, but still you can see that this is a copy. Hope we can get back something from them, atleast a ref.


Regards,
Mannu

Yep, we delt with them several months ago, they simply redesigned the user and admin interfaces, nothing else. Their 1.1.11 version code is available on the project website in accordance with the GPL license of astGUIclient.

As long as they release their versions of the code as GPL they can offer it as their own product. I should mention that they do not support the project in any way and have not donated or given back any of their sales of their product to the astGUIclient project or me. They claim that they offer their dialer as a service for customers of their telco VOIP termination services.

One thing I find interesting is that they claim SIT tone detection which is something not offered by vicidial or any Asterisk-based dialer that I know of. We all work off of Carrier signalling, not audio tone detection.

I have sent a note to Sumit Srivastava at packet-shaper asking for a code update, You can find the latest code they have sent to me on the project site here:
http://sourceforge.net/project/showfile ... _id=192630

Aheeva detects SIT tones. I know because we use it.

anyhow their skin lacks some parts. But its a good try. hope in the near future we can customize vicidial like template based.

One question though: WHAT THE HELL IS SIT?

SIT is Special Information Tone

It is used supposedly to signal that a phone number is disconnected or invalid in some way. SIT tones are different from country to country and even in the USA their amplitude or timing is not standardized even though it is supposed to be.

To analyze the SIT tone in Asterisk you would need to rewrite the Dial application and build into it a signal analyzer since the SIT tone is almost always sent before an Answer signal has been sent from the carrier meaning that another application cannot be called since the Dial has not finished.

Many dialers that claim SIT tone detection do not actually do it, they just use the standard Asterisk busy detect and/or the PRI or SIP call signalling received from the carrier. Both of these methods also have their own problems as well.

With the popularity of devices like the Telezapper and people adding SIT tones to their voicemail messages, the use of SIT tone detection has become less accurate as a form of detecting invalid phone numbers.

http://en.wikipedia.org/wiki/Special_information_tone

mflorell wrote:SIT is Special Information Tone

Many dialers that claim SIT tone detection do not actually do it, they just use the standard Asterisk busy detect and/or the PRI or SIP call signalling received from the carrier. Both of these methods also have their own problems as well.
http://en.wikipedia.org/wiki/Special_information_tone

What kind of problems are there with carrier signaling?

They are inconsistent from carrier to carrier, and in the USA there are hundreds of carriers. One carrier may send the proper signal 28 for a true disconnected number while another carrier would send a signal 31 or 34 for a disconnected number where those signals on another carrier would actually mean network congestion, not that the number was invalid so if you try a minute later you could connect to the number. The problem is noone follows the standards and they don't really care to.