We had a call using Skype conferencing today, and the experience wasn't that great for people running on Linux. It was pointed out that we should investigate using open solutions.
Here was the bulk of my post:
The Gizmo project is SIP and now works with any SIP system, not just Gizmo itself.
Jingle libraries are adding voice capabilities to various Jabber clients.
Skype works today, but we have several up and coming replacements.
I believe some folks at CivicSpace have occasional access (?) to an Asterisk box. I think we can probably fund an Asterisk box for testing out of Germany, I bet we can get one put in at OSUOSL ( http://www.osuosl.org ), and we can probably get another in at a San Fran location.
Now all we need is people experienced with Asterisk to help with defining a base setup and configuration.
Comments
Asterisk box on offer
Hi folks, I can offer up my asterisk box if you think it would help. It's hosted in British Columbia, so may or may not be suitable. But, I can be very flexible when it comes to configuration etc. It currently has only 2 PSTN lines connected, but since most people will have VOIP clients, this shouldn't be an issue.
Cheers,
Simon
Thanks for the offer, Simon
Some of this is for testing. Ideally, it would be great to interconnect a lot of these boxes Asterisk-to-Asterisk for local calling, and deploy Least Cost Routing across the whole thing making it possible to make free calls to many locations in the world. There is a lot of handwaving in that sentence, though -- I know it is possible, but don't know enough about the inner workings of Asterisk. I believe there are actually some websites dedicated to doing this as well. Someone needs to do the research on this and help come up with a plan.
In the meantime, test accounts etc. so people can work with Asterisk + Drupal would be good.
Hope to see you in Vancouver some time...great work on the OurFernie.com portal.
I'm no Asterisk expert either...
but conferencing like what we need isn't hard. The problems I have with hosting this myself is bandwidth, and NAT. I can probably only handle a couple of dozen connections, and my main server is NAT'd, which makes SIP a lot more complicated. (Although I have another Asterisk server in the UK that isn't behind NAT)
Your suggestion of asking osuosl to host this makes a lot of sense. If it would help, and I knew who to co-ordinate with from the Drupal Infrastructure team, I'd be happy to try to organise this. I'm sure that we could get a proper Asterisk expert to consult on this, because after all, the Asterisk community seems to be quite happy running on Drupal...
Oh, and thanks for the nice words about OurFernie.com - although we're still not quite up and running, we've had overwhelming support from the community.
another box availabe
hey boris,
just wanted to let you know that i have just taken over the Asterisk module for Drupal, have completed a rewrite and update to 4.7, and will be converting to 5.0 shortly.
at this point i also have an Asterisk test box going, and it can certainly be used on a small scale to help us determine the future direction of things, get other people interested, etc. this box has similar restrictions/benefits to the one simonc offered.
within the next 3 months or so, i should have a full production Asterisk box sitting in a nice datacenter, and i would certainly consider hosting some enterprise Asterisk/Drupal clients there if the details worked out.
Asterix Boxes
I have a couple asterix servers coming offline after November 7. They have been used for phone banking on a campaign and I would be happy to put them up for use on this purpose. I also have a fair amount of expertise at configuring Asterix at this point and would be happy to help out.
M