Single Step Conferencing

[Furface64]

We would like to implement single step conferencing to use as a call recording solution. We have spare station licenses and room on our Medpro boards.
There is very little information out there descibing how this is set up. How are the stations administered?
I know we use CTI to do the conference and I'm sure our programmer can figure out how to get the recording but I need to know how to set up the PBX side.

We have dual S8720s running CM 3.1.3 and AES 3.0. We also have TSAPI licensing as well.

What else will I need to implement this? Can anyone point me to a "how to" document?

Thanks

 

[Zowwie438]

This is pretty simple stuff to do!

1) Have your programmer research the TSAPI docs on single step conf.

2) Add a T1 bank to your PBX and assign vdn's to it.

3) Get your developer a list of VDNs and a dialogic board or something similar.

4) Your developer then needs the docs for dialogic to figure out what lines are busy/active, etc.

Now... Once your CTI application hears device ID 12345 throw an csta_originated or csta_initiated unsolicted message... You check teh dialogic board with code to see what port is free. Say t1 id 13330 is free.

Developer then passes tsapi command for single step conf device id 12345 to deviceid 13330 (recording line).

Then... When device 12345 throws a csta_disconnect or csta_clear... Then the recording stops.

I hope this helps. If not, I could write one for you for some $$$.

==Zow

 

[Furface64]

Thanks for the response. Could Digium's cards be used for this? We already have some of those? I'm using a couple for IVR with Asterisk right now.

 

[Zowwie438]

Yeah... I just used dialogic because that is what we use. If you wish to get really fancy, you could not use any cards and utilize h323 instead. But that would make things way more complex.

I would recommend putting the recordings in a common mp3 format due to size... plus then you could easily expose them via a webpage so folks can listen to them using their default windows media player or fav mp3 player.

Wav would just be too large.

If you wish to use linux to convert the recordings... Research LAME as well as shoutcast documentation. They have tons of tools for converting audio to certain formats and such. plus if you installed shoutcast, you may even be able to provide LIVE audio streaming of the calls. Sorta like a web based service observer.

Basically each inbound line would be like it's own audiostream or radio station. They would simply TUNE in to that live stream of the call that was in place.

More would would have to be done to stream live everything from a certain agent. But anything is possible with enough time and money.

Also... if your desks are using the VOIP phones already. The config on the phones allows them to send out an RTP stream (I think thats what it's called) whenever the handset is lifted. Developing something like that would totally elliminate the CTI portion, the t1's and the boards.


Good luck.