[OpenSIPStack] Open Sip Stack / IVR Features

[Christian W. <cwa...@gm...>]

Hi all,


long time ago I invested some time in the stack to develop a SIP based IVR.
In the past the problem was playing audio data on the client side (the file
was played to fast).

So I want to ask if there is something built in which can do:

- Media conversion to PCM
- Conferencing

Now I have some time again and I wanna ask if there is now something
In the stack which can do:

- Play / Record
- DTMF Tone recognition / generation
- Conferencing
- Dialout

I think I now got the problem I had in the past:

Since I play a file, the stack would send the data as I deliver it...
So if this is still the same, how to time the data on my side so they
a) would not be lost and b) played correctly on the other side?


Kind regards 


Christian