search for: 20letting

Displaying 3 results from an estimated 3 matches for "20letting".

2005 Aug 17
0
canreinvite in sip.conf
Hi, I'm using asterisk 1.0.6 and I would let media path be connected directly between the phones without going through Asterisk. I have to it with an AtCom320 (with pa168s chip). I just saw and tryied to do what this page http://www.voip-info.org/tiki-index.php?page=Asterisk:%20Letting%20SIP%2 0clients%20connect%20directly says. Before going on (with sniffer eth traffic between * and two phones) I'd like to known if it can works. Does anyone just did it? Thanks in advance Gio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://l...
2004 Sep 28
3
CODECs and sip.conf and voice quality
Group, Just want to share with the group my recent findings regarding CODECs/Vocoders and the effect it has had on voice quality and the intermittent noise and breakup problem I have which I mentioned in a previous emailing with the u-law CODEC. Calls again are placed through a SIP phone to a TDM400P to the PSTN. A good reference on the reasoning behind the selection of a CODEC was found in the
2004 Dec 16
3
Get asterisk out of the RTP stream?
Here is the setup: Phone A (in NYC) on own bandwidth. Phone B (in LA) on own bandwidth. Asterisk box in Houston,TX on own bandwidth. Both phones contact asterisk to register. Not much bandwidth used for this as it is a few packets every hour or so. Phone A calls Phone B. Phone A sends a call request to asterisk and asterisk calls phone B. Both phones are connected and both people are talking.