* You are viewing Posts Tagged ‘amount’

Total Amount Of Asterisk Installations

Counting any Open Source package is difficult for many reasons. There is probably not a reliable answer to this question since there are at least 4 major “flavors” of Asterisk out there (1.4, 1.6, 1.8, 1.10) and open and commercial source. It is reliably > 10,000 and quite possibly over 100,000 or even over 1 million. The Asterisk folks might be willing to tell you how many downloads have been done from http://www.asterisk.org , but that wouldn’t tell you the real number.

Maybe a good start point for an estimate would start at 200,000+ if you are including all of the versions and types. But then we might still think about the Asterisk boxes that are plugged to the Internet.

Getting a reasonably accurate count maybe would not be that difficult, but everybody is so paranoid about anybody knowing anything about them and what they do.

Some community members, like Danny Nicholas, points out the idea of a ‘curl’ request in the script that starts Asterisk that sends your MAC address and Asterisk version number to Asterisk.org. Personally I think that’s a great idea, as there’s no IP address tracking involved or any other identifying information, just the MAC and cheese. Another important remark is that, being Open Source, you can see exactly what is being sent and could always ‘opt-out.’

Some really useful information could be gathered and displayed like:

  • ‘Popularity’ of different versions.
  • Average time between restarts by version number.
  • Ratio of starts to stops by version number. (The difference between starts and stops could be an indicator crashes.)

Other information that might be helpful to share would be the TDM capacity or maximum simultaneous call count. And all that without really getting ‘compromised’ regarding the shared information. After all, what ‘competitive advantage’ would someone have over you just knowing that Asterisk was started on a box owned by someone, somewhere?

app_swift tts module – new home.

Hi Folks,

After receiving a surprising amount of emails from Asterisk community
members, I thought I’d fire something off to the users list to clear
any confusion regarding the Asterisk Forge (forge.asterisk.org)
website and the future of the app_swift text-to-speech module.

With regards to the Asterisk Forge website redirecting to GitHub, this
has been a long time coming. Emails were sent out to the various lists
warning folks that the hosted GForge site was going away – so no one
should be too surprised – ‘nuf said there.

As far as the app_swift project is concerned, with the exception of
moving things around as far as location, it is business as usual.

The app_swift code for *all* the different versions of Asterisk is now
being hosted on GitHub at https://github.com/dmsessions/app_swift .
This is a good thing and will make life easier.

btw, I love git. If you don’t yet, you will . . someday soon . .

Individual tar files for each of the different versions of app_swift,
which is what 99% of people are going to want, are all available for
download on my website at http://www.darrensessions.com by clicking
the ‘Downloads’ button at the very top of the page.

That is all my friends.

Seasons Greetings!

– Darren

Progress, Delay, DTMF / background calling

Hi,

has the following been done before respectively is it possible with
Asterisk? I searched the archives but couldn’t locate anything.

1. Call to 5555 comes in via SIP.
2. Call is not answered yet but progress continues.
3. At the moment the call comes in something like this gets spawned in the
background:

Dial(SIP/123456@provider,,D(ww${EXTEN})
which should translate to:
Dial(SIP/123456@provider,,D(ww5555)
But even better would be take the ${EXTEN} and put some w’s between them:
Dial(SIP/123456@provider,,D(ww5ww5ww5ww5)

4. After a pretermined amount of time since the call came in respectively
the Dial command was spawned “in the background”, e.g. 15 seconds,
Asterisk answers the call and the call legs are connected together.

So, with some fantasy commands, something like this:

_X.,1,Progress
_X.,2,DialBackground(SIP/123456@provider,,D(ww${EwwXwwTwwEwwN}),ANSWER-AND-CONNECT-LEGS(15)

I hope my request is not too cryptic. In short: I’d like to receive calls
to arbitrary extensions, but not answer them directly, only after a Dial
command has been spawned and a predetermined amount of time has passed
since the Dial command has been spawned / since the Dial command has
connected to 123456.

Possible?

I’m new to the list, hi! :)

Thank you!

Any way to temporarily disable a registered SIP PEER in Asterisk?

Hi Everyone,

We want to be able to momentarily or temporarily provide CONGESTION or
DE-REGISTER a SIP PEER to asterisk through WEB GUI. I don’t want to indulge
into dial-plan and write changes to .conf file every-time. Is there any way
that a SIP PEER can be de-registered for an amount of time or maybe
deactivated? or there isn’t such facility available in asterisk?

Thanks

Mail list Woes?

Anybody notice log delays in this list, and very small amount of traffic?