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 , 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…

General 3.3 years ago 0 Answers

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 (
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…

Asterisk Users 3.9 years ago 0 Answers

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:
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…

Asterisk Users 4.5 years ago 0 Answers

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

Asterisk Users 4.6 years ago 0 Answers