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
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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.