David,
What someone is missing, is that TDM and VoIP aren't "converted." TDM PRI's include signaling in the same bitstream. VoIP uses separate data paths for signaling and voice. The voice data can be the same, or different.
It is my understanding that TDM is circuit switched and VoIP is packet switched. It would seem to me that at some point in a TDM-VoIP gateway, a change from circuit switching to packet switching is happening, and vice versa depending on the direction of the signals. I was just wondering if anyone could detail that process and tell me if it is resource-intensive. If I'm completely off-base, please point me in the right direction.
Agreed. But not to him. T1 refers to the line coding on 2 physical pairs of wire to encode and carry a 1.544 Mbps datastream. T2 is four of those signals multiplexed onto 2 pairs of wire. T3 is 28 DS1's (7 DS2's) multiplexed onto two coaxial cables. DS1 is a logical concept that defines what to do with that signal, what the bits mean; each DS1 is made up of 24 DS0 time slots. An ISDN PRI is the definition that one uses 23 of the time slots for 23 voice channels, plus one DS0 dedicated to signaling. A DS3 is 28 DS1's (and is usually carried on a T3 physical layer, does it begin to make sense?)
Yes, thank you.
It's too bad. A lot of people without any telephone background try to make up stuff using pieces of the old terminology and wonder why they stay confused. They could look it up, but they don't. For instance DID's. DID has a specific meaning and inward service from the PSTN handed off on VOIP isn't it.
Do you have a good, reliable source that I could take a look at?
No. It writes data to whatever format the sound card supports, usually 16 bit linear (raw) which becomes .wav if you add file headers to it.- What codec does the Monitor application use when digitally recording
calls (if possible, I would like to avoid transcoding the streams when
recording and let sox handle the conversions on a different box)?
I *believe* that it will write the data in G.711 format. Don't rely on this though.
Is there a way to specify the format? What if there is no sound card on the Asterisk server?
Thank you,
Matthew Roth http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian _______________________________________________ Asterisk-Biz mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-biz
