VoIP Mailing List Archives

[msc at freeswitch.org]

Ladies and Gentlemen,

Thank you for calling in to the weekly FreeSWITCH conference call. Last week's agenda was rather light, so if you have things that you would like to have discussed please be sure to add them here:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_09

Here are a few updates for everyone to keep in mind:
* Starting with the upcoming meeting (Oct 9) the conference will support 48kHz CELT codec.
* When you call in, please mute your phone if you are just listening or if you will not be speaking a lot. Background noise from one user isn't usually too bad, but when 10 or 15 people are not muted it can get a little distracting.
* We are always looking for people to help out. If you are looking for ways to help out then please by all means call the conference. If you cannot call the conference for some reason but still want to help out, please email me off list. We have numerous documentation, janitorial, code review, etc. sub-projects that FreeSWITCH users can help with.
* Be sure to update to latest SVN trunk and test test test! We are prepping for 1.0.5 and more people testing means a more stable release delivered more quickly.

One topic that came up was the testing of Mike van Lammerman's Ultramonkey setup. (http://bit.ly/8zZUS) Several people agreed to try it out. We would love to see others try it out and report back their experiences.

Lastly, we'd like to thank everyone who has been helping out with testing, documentation, and answering questions on the email list and IRC channels. A special thanks to Diego Viola for doing lots of wiki cleanup lately. Everyone's efforts are appreciated.

-Michael

[brian at freeswitch.org]

It always supported 48kHz CELT but the conference itself was running
at 32kHz so everyone 48k had to be down sampled. Now you all get to
be up sampled. w00t!

/b

On Oct 5, 2009, at 2:03 PM, Michael Collins wrote:

[shaheryarkh at googlem...]

I got a lot of problem last week for making conference call. I was at home (conference call starts at 2200hours PKST, my time) and unable to make SIP call since the government has blocked it. So my only choice was Skype, but unfortunately DTMF wasn't working, i get connected on Skypiax5 for about 7-8 times, but DTMF problem never let me get in to conference. Is there any solution for this?

I really need to discuss a lot of things about FS documentation in conference, like what FS community can expect from it and what not? how i have planned it? how much progress has done? what are the problems me and my team are facing (which has slow us down considerably)? etc. etc.

I have a solution for it but that needs testing. The plan is to use one of my FS servers to connect my jingle calls from GTalk to conference server over SIP. How can i test this setup with conference server, any ideas?

If any one else also interested in getting connected to weekly conference call through this setup then i can also extend this setup as needed.

Thank you.


On Tue, Oct 6, 2009 at 1:30 AM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[chris.chen2004 at gmai...]

Hi Muhammad, the simple and reliable solution for you where SIP is being blocked is add conf+888@conference.freeswitch.org ([email]conf%2B888@conference.freeswitch.org[/email]) to your Goolgetalk buddy list, and you can call from there to join the conference, simple and straightforward.

Chris

On Wed, Oct 7, 2009 at 12:24 AM, Muhammad Shahzad <shaheryarkh@googlemail.com (shaheryarkh@googlemail.com)> wrote:

[shaheryarkh at googlem...]

Great. Just added it. Is there any user limit on this?
 
Thank you.


On Wed, Oct 7, 2009 at 6:15 PM, Chris Chen <chris.chen2004@gmail.com (chris.chen2004@gmail.com)> wrote:

[brian at freeswitch.org]

No!

/b

On Oct 7, 2009, at 8:37 AM, Muhammad Shahzad wrote: