rtp bugs in Asterisk

But I think I fixed DTMF sending on our VOIP server. DTMF is the sound that you hear on your phone when you press a number button. In a normal phone world, this sound is made by your phone and sent across the line. When you work in a Voice over IP world, it's not so black and white. You can do it the same way as a traditional phone - this is called "in band" DTMF. This doesn't work very well when you compress the sound for transmission over the net. More commonly we use RFC2833 which transmits the DTMF signal using a message.

Anyway - to the point. I found a bug where the DTMF signal wouldn't be sent. Basically, when it sends it it is supposed to include a time stamp that tells the remote computer where in the sequence of the life of the world the dtmf signal probably should be. Unfortunately, this was (for some strange reason) -96, which meant it didn't work. I hunted and hunted and discovered that this magical time stamp was never really set properly. When I set it, it all of a sudden started to work magically.

Now, provided my fix doesn't break anything I can submit my first patch to Digium for Asterisk. That's a little exciting :-) (even if it is only for one line...)