Digium has partnered with VoIPSupply to hold a contest for coming up with a list of things that can be done with Asterisk.
If you have an interesting application of Asterisk, be sure to add it to the list!
http://blog.voipsupply.com/asterisk-news/101-things-you-can-do-with-asterisk
Russell Bryant 
Send out mass ‘wake up calls’ to all my friends because by Golley if I have to be up at 5AM so should they!
Things I would like to see in asterisk:
1- Better core.
2- XML / XMLRPC / yaml / normal/plain text configuration.
3- software based conference, with the ability for dialing out, etc.
4- GPLv3 only, so I don’t have to give my credit to Digium, and they can still sell the software as free/libre open source software.
5- Better collaboration, better development model, git?
6- All and better features than FreeSWITCH. http://wiki.freeswitch.org/wiki/Specsheet
I hope you don’t take this in a bad way, is not meant to troll or create a flamewar, this is just my point of view, and I think with these things asterisk will outstand anything.
7- Get rid of chan_sip.c and implement Sofia-SIP, or make the SIP stack in Asterisk full RFC compliant with ALL the features.
8- The ability to dial from the cli, without having to put a extension in the dialplan first.
If you don’t provide this things in asterisk, I guarantee you that most users are going to start switching to FreeSWITCH.
1. The core is constantly improving. For example:
http://www.russellbryant.net/blog/index.php/2008/02/18/request-for-testing-distributed-device-state/
http://www.russellbryant.net/blog/index.php/2007/12/31/teamrussellchan_refcount-improving-asterisk-performance/
2. Well, obviously we already have plain text configuration. We also have the realtime configuration interface. In Asterisk 1.6, there is a curl realtime engine. This allows Asterisk to make HTTP requests to retrieve configuration and you can write your configuration engine in whatever language you want on the web server. However, providing a socket interface that you can connect with and change any part of the configuration would be great. It is something that we have talked about for a long time and will do when we rework the entire configuration interface.
3. This is already done and will be in future releases of Asterisk 1.6. It is in Josh Colp’s bridging developer branch.
4. Without getting into the details of licensing, you don’t give credit to Digium. You retain copyright of your code. For more details, see the following blog posts by Digium’s CEO:
http://blogs.digium.com/2008/04/25/asterisk-myth-busters-episode-2/
http://blogs.digium.com/2008/05/13/asterisk-myth-busters-episode-3/
5. A better development model? You may need to be more specific. git is a tool, not a development model.
6. Most of those things already exist. Many others are currently in development. In addition, Asterisk has many things that FreeSWITCH does not. However, there will always be differences. Use what works best for you.
7. This is another blanket statement. Which features are you referring to, specifically? SIP TCP and TLS are already supported, and SRTP is coming soon.
8. You can already do this using the “originate” CLI command.
Cool, thank you.
You are doing an awesome work with asterisk.
Keep up the good work.
I wonder if what this guy says about Asterisk is still true:
http://digg.com/software/How_does_FreeSWITCH_compare_to_Asterisk
See, I really like asterisk and if that’s the case I would like to see it improve, it is true that asterisk still depends on some modules to be able to start? is asterisk module independent now?
I don’t have any reasons to switch to FS, I’m used to asterisk and I like it, I see is getting better and better as time goes by… and I know 1.6 will rock as it gets more features and solid core.
Thanks for your hard work.
If you rework the configuration interface, please don’t get rid of the text configuration, I love asterisk configuration, I think plain text configuration is really fast, it could get better though…
That’s one of the things I don’t like about FS, that only XML is supported… and they lack some applications.
Asterisk is a lot better on this.
>5. A better development model? You may need to be more specific. git is a tool, not a development model.
You are right, Git is a tool… but it’s a very innovative one, it’s based on distribution (distributed development model).
I think you will really like it, and I don’t meant to force you on using it, but you might want to check it out, it has some advantages.
Look this video 😉
Git rocks!
> 7. This is another blanket statement. Which features are you referring to, specifically? SIP TCP and TLS are already supported, and SRTP is coming soon.
I know you guys are doing the best to make the SIP stack better in Asterisk, adding new features, etc. and I’m really happy of that, but do you guys aim for RFC compliance?
“I know you guys are doing the best to make the SIP stack better in Asterisk, adding new features, etc. and I’m really happy of that, but do you guys aim for RFC compliance?”
Yes, of course we do. However, that’s not _all_ that we aim for. In fact, our #1 priority is making it work with reality. In reality, a lot of SIP implementations do weird things. We make chan_sip act in weird ways sometimes to get the best possible interoperability that we can. We have been extremely successful in that regard.
“I wonder if what this guy says about Asterisk is still true:
http://digg.com/software/How_does_FreeSWITCH_compare_to_Asterisk”
Most of that is not true anymore. It’s obvious to me that it was written by someone with knowledge about Asterisk 1.2 and earlier. A _lot_ has changed since then. Writing a detailed response to that article is on my to-do list.
“See, I really like asterisk and if that’s the case I would like to see it improve, it is true that asterisk still depends on some modules to be able to start? is asterisk module independent now?”
No, Asterisk does not depend on any modules to start. The ones that were required have been moved into the core, which was only one of them (res_features). In Asterisk 1.6, that code has moved into the core.
Thank you for taking the time of answering my questions.
I really love asterisk and I think it’s progressing rapidly.
I also look forward in your response to that guy =D
Keep up the incredible job! =D
a socket interface will indeed rock… do you guys have plans for doing this in 1.6?
if you rework the entire configuration interface please also allow regular expressions with PCRE for dialplans, etc.
Russell, could you please post an example dialplan to demonstrate the prank you describe on the 101 things page where you connect two external calls with each other’s caller ID’s?