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 wouldnt 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…
I have an internal extension, e.g. 1005 which is being called from an
external/public number like 123456789. Now when it comes to the spoken
voicemail information it says something like "number 1000 not available",
however it should say "number 123456789 not available". How can I configure
this? I already googled and I guess this is really easy, but I just
couldn't figure out how to do this ): So thanks for any hint :-)
Can you set up asterisk so when a 911 call is placed, in addition to the
call out to the PSAP, it also alerts multiple other phones on the switch
and will display detailed information. Such as alerting a receptionist
or security guard there is a 911 call elsewhere in the building and the
location of that call within the building? If so, how? Thanks in advance for the help...
I'm new to asterisk, currently working through Asterisk, The Definitive Guide. I have a couple Grandstream 200 phones and a Polycom 501 to play with. The Grandstreams were very easy to configure, but the more capable Polycom almost drove me crazy. There is so much conflicting information on the web about necessary config files and names of those files. Anyway, all my phones are now working on a basic level, but it occurs to me that once spend a lot of time learning Asterisk, I might then have to spend an equal amount of time figuring out how to configure…
sorry if this is a silly question. My recharge application uses * digits if the subscriber wants to send
some aditional information to speed up a process, dialing something
On my old ss7 network works great, but on my new ngn/sip i think it's
not possible because somewhere the call is rejected.
-On the NGN/Ericsson side engineer say that the call whas deliverd.
-On the asterisk side there is no invite shown on debug.
Can sip one or more * signs in a dial?
Can someome give tested and proven information on Asterisk capabilities?
I have a server with 24 cores running at 2.4ghz and 16 GB RAM. How many concurrent SIP sessions I can run from single instance of Asterisk on this server? I wish to use G711 codec with echo cancel. And all calls needs to be recorded.
What will be impact on no of session when G729a is used?
Thanks & Regards,
I know that I can use the AGI to call (run) a script (php or python or any other kind of scripts), but the question is:
If I have information that I need to build a decision in the extensions.conf based on it, and these informations can be obtained using this script, so how I will read these informations? What is the method to read it from the database and store it in a variable that I can use it in the extensions.conf to do proper call routing? How?
I'm seeking information on how to report an IP phone
on a call that is occurring on another IP phone. Example: While the A phone is ringing, Asterisk sends a
notification to a phone B on the call that is going to A, but this
notification is displayed on the B phone display and the user does not
need to hit anything to view the information. I'm working with "Siemens optiPoint 410 Economy" and
"Yealink T22P" phones. I Have searched some information on the Web, but without many
details and solutions. I…
Hi Team, Does any one know how to set IDLE FEATURES on the Avaya 4610sw IP Phone when it registers directly on the Asterisk Server ? I have done many things like setting it through the 46xxsettings.txt but it doesn't work: SET IDLEFEATURES "326" SET FBONCASCREEN 0/1 ;;; ( did both 0 and 1 but doesn't matter what it is, the idle screen doesn't show up the voicemail button ) It would be great if someone could help me out in this regard. Thanking you in anticipation. Regards, Aamir Chougule Confidentiality Notice: This e-mail, including any attachments, may contain information…
I'm reading some information that recommends using SER / OpenSER for
large installation to offload SIP traffic from the Asterisk server. http://www.voip-info.org/wiki/view/Asterisk+at+large However, it looks like the information might be dated. I'm looking at a potential 750 SIP phone and 150 Analog installation,
all internal network, PRI trunks, and am trying to nail down an
architecture. Opinions? You think I skip the SER box if I'm using 1.8? Thanks!