Here are some more videos from the last ClueCon<br><a href="http://files.freeswitch.org/cluecon_2008/">http://files.freeswitch.org/cluecon_2008/</a><br><br><div class="gmail_quote">On Thu, Oct 23, 2008 at 5:03 AM, Thomas Troesch <span dir="ltr"><<a href="mailto:ttroesch@dslextreme.com">ttroesch@dslextreme.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Thank you for your input. I am clearly naive about what may be required,<br>
and I need to get more details on what is currently in use.<br>
<br>
The environment is not 'can't-be-down-for-a-second' - it has crashed<br>
before - but that became an emergency. It is possible to plan for short<br>
( 1/2 hour ) downtimes, but not easily.<br>
<br>
Thank you for the link - it was interesting. I also listened to some talks<br>
by Diana Cionoiu of YATES and Nenad Corbic from Sangoma that I found on<br>
you-tube ( from Asterisk-Tag in Germany ). It helped by giving some<br>
context for things I'm not familiar with.<br>
<br>
I'll check in on the IRC when I have some questions.<br>
<div><div></div><div class="Wj3C7c"><br>
On Wednesday 22 October 2008 15:01:34 Michael Collins wrote:<br>
> On Wed, Oct 22, 2008 at 9:50 AM, Thomas Troesch<br>
<<a href="mailto:ttroesch@dslextreme.com">ttroesch@dslextreme.com</a>>wrote:<br>
> > Is there a document or recommended strategy on how to migrate from an<br>
> > existing PBX system to FS? There are 2 T1-PRI lines used 24/7 (call<br>
> > center) which can't be down for development or testing. The existing<br>
> > system is InterTel.<br>
><br>
> Right now the OpenZAP stuff is still in development. I can't recommend<br>
> putting it in a 24/7 can't-be-down-for-a-second kind of environment, at<br>
> least not yet. We're getting there... Also, be sure to note which PRI<br>
> protocol variant (aka "dialect") you are currently using. OZ supports DMS<br>
> and 5ESS pretty well but NI2 is still a little iffy. Again, we're working<br>
> on it but there are only so many hours in the day...<br>
><br>
> As for documentation, I'm almost positive that there aren't any docs<br>
> extant that describe this process. I do know that Yossi N. has migrated<br>
> from Asterisk to FreeSWITCH and has some experience. His presentation at<br>
> ClueCon this year is available here:<br>
> <a href="http://files.freeswitch.org/cluecon_2008/Day_03.Presentation_11.Yossi_Nei" target="_blank">http://files.freeswitch.org/cluecon_2008/Day_03.Presentation_11.Yossi_Nei</a><br>
>man.1500kbps.mp4<br>
><br>
> However, I don't think we're at the point of how to rip out a legacy PBX<br>
> and drop in FS. Not yet, but give us some time... :)<br>
><br>
> > I have a Sangoma A104D available with a tap ready to be installed. I'm<br>
> > thinking (hoping?) that I can start by using it to monitor and/or<br>
> > record calls without interfering with the InterTel.<br>
><br>
> I am interested in knowing if you can pull this off. I'd love to see this<br>
> in action. I've got an A104D and I would be very interested in seeing<br>
> someone get the HI-Z setup nailed down so that you can tap T1 lines. I'm<br>
> particularly interested in sniffing d-chan traffic but I would also be<br>
> interested in seeing someone use FS to create a call-recording system. (I<br>
> know that Tri-Sys in NJ uses YATE + Sangoma for this, so I'm 100% sure<br>
> that it is possible.)<br>
><br>
> If you feel like being a trailblazer then by all means go for it! Hop on<br>
> IRC (#freeswitch and #openzap on <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a>) and pepper us all with<br>
> questions. The only rules are "be cool" and "document it on the wiki if<br>
> you learn something that wasn't already there."<br>
><br>
> -MC<br>
<br>
</div></div><div><div></div><div class="Wj3C7c">_______________________________________________<br>
Freeswitch-users mailing list<br>
<a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
</div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
<br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="http://iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a><br>
<a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:213-799-1400<br>