Re: VOIP concall with the Freeswitch folk 1PM EST Mar 9

[d@taht.net (Dave Täht)]

Concall at 1800 UTC (1200 CST/1PM EST) (roughly half an hour from now)

sip:888@conference.freeswitch.org or via the good old PSTN at +1-919-386-9900 or +44-3300-100-295 

gtalk:conf+888@conference.freeswitch.org Or click on this link 
Or call Skype the skype user "skypiax5", you'll be automatically connected (max 20 concurrent users). 

Codecs: PCMU/PCMA, G.722, CELT, Speex, Skype, among others
Press 0 to mute/unmute your self. Press * to deaf/undeaf. 

To get the best sip quality:

If you use sip already, just dial the 888@conference.freeswitch.org
number....

if you don't already use SIP... 

you can setup an account on your sip client (I'm using linphone) to use
IPv4, send DTMF as sip info, put in your "behind nat public IP address",
enable echo cancellation...

create the account of sip:your_name@yourdomain, AND use 

sip:conference.freeswitch.org as the sip proxy

and then dial  888@conference.freeswitch.org to join the conference.

I'm connected using speex32 and it sounds pretty good. 

More details on the call are at http://www.bufferbloat.net/news/8

There will also be chatter on the irc channels of #bufferbloat and
#freeswitch on irc.freenode.org


d@taht.net (Dave Täht) writes:

> Members of the bufferbloat community have been invited to open a dialog
> with the freeswitch[1] and VOIP community in the regular conference
> call, 1PM EST Mar 9.
>
> Details on the conference call, dial-in methods, tools, etc are at:
>
> http://www.bufferbloat.net/news/8
>
> (I sure hope I got the time right. If I got it wrong, I will correct it
> on the news page)
>
> Hope to hear you there! (I also expect much chatter in irc)

-- 
Dave Taht
http://nex-6.taht.net