By davec611 on

Forest lake Online(www.forestlake.ca) is a website dedicated to a cottage community in the Canadian Laurentians. We recently added the ability to make phone calls between members directly from the site using the asterisk module developed by Malthus. If you are interested in trying it out, go to the site, register (you can automatically register with your drupal id and password from another Drupal site), and follow the instructions provided via the 'FL Tel' link in the top menu. Let us know your thoughts and comments!

There is still some annoying bugs, but the site basically works. We intend to improve it as time permits. If you are interested in Drupal-Asterisk Integration, Malthus has posted a suggested roadmap elsewher on drupal.org which gives an idea of what needs to be done. I am sure he would appreciate any help offered.

I would like to thank Malthus for his patient assistance in helping me get this working. I would also like to thank Bill Lloyd of www.prodosec.com for setting up the Asterisk side. Without their time and effort, this would have been an impossible task.

Davec
Forest Lake Online

Comments

davec611’s picture

davec611 commented

I forgot to add a few links. If you are interested in trying out asterisk telephony, the module can be found at http://drupal.org/project/asterisk . It links two great open source developments, Drupal and Asterisk

Malthus's suggested roadmap for asterisk can be found at http://drupal.org/node/37571 .

You can check out asterisk itself (notice it is a Drupal site) at http://www.asterisk.org/ .

Finally a more direct link to the Forest Lake Telephone Company instructions can be found at FL Tel . When you get there and register you can use the box in the upper left corner to phone anyone in North America.

Feedback is welcome

Davec

Forest Lake Online

KSA213755’s picture

KSA213755 commented

Dave, that's a real nice site. The Asterisk integration is really cool. I didn't understand exactly how it worked until I read the post on your site explaining the process. I have a project I'm working on that could really benefit from the asterisk integration, so thanks for showcasing FL Online as an example of how it can be done.

Bug: You may already be aware of it but the calendar in the left sidebar on the events page flows over into the content section in Firefox. It works fine in IE6.

Roger

davec611’s picture

davec611 commented

Thanks Roger.

I am going to offer my asterisk server as a test bed for anyone who wants to try the asterisk module, so when you get closer, let me know. ( It will likely only be for soft phones unless I can figure out a way to recoup the PSTN charges)

In the meantime feel free to try a call on my site.

Re the event calendar, I had that bug fixed in 4.5 but had to do the upgrade to 4.6 (major job ) in order to use asterisk. I just need to find a moment to fix it again

Davec

Forest Lake Online

KSA213755’s picture

KSA213755 commented

I tried it out and it worked great.

Do you have your Asterisk server connected to a POTS lines or are you using a VOIP service like Broadvoice?

Also, I've been unclear as to exactly how Asterisk works in terms of multiple calls. For example, say you have a single line to access the PSTN network. A person places a call that goes out over PSTN and is connected to the party they are calling. While that call is taking place, is the PSTN access tied up, or does another called still have the ability place a PSTN call? I assume that multiple IP based calls can be handled simultaneously.

Roger

davec611’s picture

davec611 commented

Roger, I am using a VOIP service called appropriately enough Voipjet ( http://www.voipjet.com ) which can handle as many calls as you like and connect them to the pstn. It's base rate is 1.2 cents per minute within North America. Bear in mind when yoiu are using asterisk integration and initiating calls from a website, you are actually making two calls (ergo 2.4 cents). It is very simple to set up for these purposes anyway.

If you want to interface directly to the pstn, you need one line per call which usuually means a channel bank I think.

Davec

Forest Lake Online

davec611’s picture

davec611 commented

Roger, I am using a VOIP service called appropriately enough Voipjet ( http://www.voipjet.com ) which can handle as many calls as you like and connect them to the pstn. It's base rate is 1.2 cents per minute within North America. Bear in mind when yoiu are using asterisk integration and initiating calls from a website, you are actually making two calls (ergo 2.4 cents). It is very simple to set up for these purposes anyway.

If you want to interface directly to the pstn, you need one line per call which usuually means a channel bank I think.

Davec

Forest Lake Online

KSA213755’s picture

KSA213755 commented

So, if I understand this correctly, with this kind of setup, you can have numerous calls taking place through the site simultaneously with all of them being billed at 1.2 cents X 2 per minute. At least so long as the VOIPJet simultaneous connection rules are followed. Is that correct?

At this time, it looks like the Asterisk integration hasn't been upgraded to 4.7. The site I'm working on where I would want to use this integration is using 4.7. By the time I get the site ready to test, maybe Asterisk will be available for 4.7.

Roger

davec611’s picture

davec611 commented

Roger, There is no theoretical limit to the number of calls. It is a question of capacity of the asterisk server. Unfortunately that is somewhat limited by the current architecture of the asterisk integration module which is set up to poll the drupal sites. The roadmap that Malthus proposes ( see above links) would fix that problem and make the process much more efficient.

Even in it's present state, I understand that it should be quite feasible to initiate three or four calls a minute, even on my small server. Since Asterisk is fairly scalable, it then becomes a function of how much horsepower you want to throw at it.

I am not sure what the plans are to upgrade asterisk to 4.7, nor do I know whether it is trivial or not. You might want to ask malthus what plans there are. He can be reached via his profile at http://drupal.org/user/4932

Davec

Forest Lake Online

davec611’s picture

davec611 commented

The Forest Lake site got some nice compliments from www.brianpuccio.net for its integration with telephone technology. The key comment was that FL Tel is a "a shining example of how to tie together really cool technology the right way"

Thanks Brian. We appreciate flattery as much as anybody. Drupal and Asterisk make a great combination.

If you are a Drupal community member and want to try FL Tel, please go to www.forestlake.ca, read the FL Tel instructions in the top menu, and try it out.

Davec

Forest Lake Online

ajwwong’s picture

ajwwong commented

Great stuff. Exciting ideas.

This is friggin' amazing.... Stunning... truly...

.... Here's some use feedback.. just fyi...

OK, so... when I tried to "call user" ["Dave" -- and also my cell phone -- as the "destination phones"] here's what happened:

I tried calling user "Dave" twice. I got my [initiating phone] to ring and then attempt to connect to the [destination] phone. However, instead of a connection, I got the message "goodbye"

I tried calling my cell phone three times. Once again, the [initiating phone] rang, and I could verify that the [destination phone] always rang.

The first time this occurred -- I let it roll over to voicemail on the destination phone, but instead of a connection, I got a "goodbye".

The second time this occurred -- I picked up my cell phone [destination] -- I got a connection.

The third time this occurred -- I let it roll over to voicemail on the destination phone, -- and I got a connection.

anyhow. Neat stuff.

I'm trying to set up a 24 hour crisis hotline and this would be great. Do you have any idea about what's the cost of setting up asterisk?

Thanks,

Albert
www.ithou.org

davec611’s picture

davec611 commented

Thanks for the comments Albert, and thanks for your feedback. The system always seems to make a connection when someone picks up (i.e. an off hook situation), but is variable when it goes to voicemail. Being cheap I use an answering machine, and I thought that was the problem, but it also does not work necessarily with voice mail. ( One exception - I have a friend with a Vonage account and it will always connect to his voice mail.)

I am fairly sure that anyone with more Asterisk knowledge than me (or even telephony knowledge) could diagnose the problem and point out the solution. So if anyone reading this knows what to do please let me know..

I have been out of the country since shortly after launching this, and have not had time to look into a solution nor the means to implement it. But I will get on it when I get back to my home base.

With respect to asterisk setup, a friend of mine loaded my machine (which needs to be dedicated). I have seen people advertising a fully loaded asterisk system including the box fo anywhere from $1500 to $3000 on www.voipinfo.org, so that gives you an idea. If you are technical, self installation does not look too bad, but you then need to understand asterisk setup.

As I think I mentioned above, I am willing to allow registration on my asterisk box for testing purposes, so if you get to that stage, let me know and I will try to help.

Davec

Forest Lake Online

davec611’s picture

davec611 commented

Albert,

The non connection problems seem to have been caused by an unusually short time delay. I changed the wait time from 20 to 30 seconds, and now virtually all calls seem to go through to the PSTN voicemail or answering machine, if they are not picked up by the user.

If you have a chance please give it another try at http://forestlake.ca/forestlake/?q=node/466

Regards

Davec

Forest Lake Online