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Asterisk VoIP Software 12.0.0-beta2 Now Available!

The Asterisk Development Team is pleased to announce the second beta release of Asterisk 12.0.0. You can immediately download this release at http://downloads.asterisk.org/pub/telephony/asterisk/releases

We strongly encourage all interested Asterisk users to participate throughout the testing process. For any issues you might find, please use the issue tracker to report it: https://issues.asterisk.org/jira. We would like you to come to the #asterisk-bugs channel in order to help communicating issues you found. Also, it is also very useful to see successful test reports. You can use the asterisk-dev mailing list for that (http://lists.digium.com).

The next major release in the series of our favorite VoIP software will be Asterisk 12, which will be a Standard release just like it was Asterisk 10.

There are many new features included in this version of Asterisk, besides of a long list of improvements. Just to mention some of them:

  • A new SIP channel driver and accompanying SIP stack named chan_pjsip has been added.
  • The Asterisk REST Interface (ARI) has been added.
  • Major standardization of the Asterisk Manager Interface and its events have occurred within this version.
  • All bridging within Asterisk is now performed using the Asterisk Bridging API, which previously was only used by the ConfBridge application.

And the list continues. For information about the new features, please visit the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Documentation

Thank you for your continued support of Asterisk!

Moving Calls From One E1 To Another

Hello everyone. I want to migrate an old PBX which uses the Е1-PRI from one Telecom to VOIP by transparently moving the numbers one by one. I mean that the numbers that the PBX handle must be transparently moved from one operator to another. The old connection to the PBX is Е1-PRI and we must preserve that because no one knows how to configure this PBX. So my idea is to connect a PC with 2 ports E1 module between the PBX and the old telecom. One port of the module will be connected to the telecom’s wire and the other port will be connected to the PBX. This PBX will be powered with Asterisk of course so it will be able to connect by SIP
trunk to the alternative VoIP telecom. So my question is will I be able to transfer the calls from one E1 port to other? Could I be able to specify the difference clock source for the both ports?

Thanks in advance. Dimitar

Asterisk Sizing For Play And Dtmf Detection

Dear all

i’m planning a migration to asterisk for a high volume IVR service
(from 1000 to 1500 concurrent call)

The IVR service is based only on DTMF tones so the features required is

- play feature
dtmf detection

Asterisk will receive calls via VOIP (SIP with g711 codec)

The IVR service wil be a static service based on Asterisk dialplan with some prompt (from 0 to 5, play of files in the same codec of the received call) and some dtmf detections.

How many simultaneous call can i handle per server? each server will have:

4 core 3.0 Ghz
4 GB of RAM

I need an aproximate sizing:

0-100 calls per server ?
100-200 calls per server ?
200-300 calls per server ?
300-400 calls per server?
400-500 calls per server?

Thanks to all in advance

Playing Music Through VoIP Handsets While On Hook

This is something I’ve seen with some key systems and PBXs. When the phones are on-hook, they can play music throughout the office instead of having an overhead speaker system do it. Never heard of it being done with VoIP, but figured I’d ask if anyone else has. I don’t see any way to do this on any phones I’ve looked at.

Need Help Designing Implementation


I’d like to replace my current VOIP provider with an Asterisk based solution. I have some ideas I want to run by the list to see if they are possible, and get answers to a couple questions.

I want to setup two Asterisk servers that are linked to each other:
– The first server would be my “external” (public) server and would live in a real data center. The second server would be my “internal”
(private) server and would live in my house.
– The external server would receive all incoming calls and handle the voice mail stuff.
– The internal server would run all the phones in my house (VOIP or Analog-via-FXS). All outgoing calls would be routed out through the external server.

I also want to add the following additional functionality:
– If the external server looses connectivity to the internal server while a call is in progress, the external server should place the call on hold while it tries to reach us via our cell phones. A message should be played informing the remote party that the connection had been lost and it is trying to re-establish it now. If it can’t reach us, it should inform the remote party that the connection could not be re-established and allow the remote party to leave some closing remarks on the voice mail system.

- If a call comes in and no one is at home to take the call (or if all lines at home are busy), it should ring all of our cell phones and whoever answers the call first gets the call. If no one answers the call via the cell phones after 3 rings, it should route the call to the voice mail system. I say 3 rings on the cell phone because I do not want the cell phone voice mail to take the call.

- I also would like the system to automatically route all calls directly to voice mail depending on the time of day (say 10PM to 8AM).
I would like specify in a “white list” specific phone numbers that are allowed to ring through regardless of time of day (i.e. her parents, my parents).

- I would like the VOIP phones to turn on the voice mail waiting indicator light if the external server has new voice messages.

Is all of this possible? If not, which part’s are not (and how much work do you think would be needed to make those parts work)?

Fax Configuration

What is the best way for me to setup Fax Capability with VOIP only.

I have a Asterisk Server hosted on the internet without a modem. I’m using Flowroute, which is working awesome, for VOIP calls.

I only have a SIP Phone at home and two Printer/Scanner/Fax Printers.

I’m not sure which Fax Addons or Extensions I should use for Asterisk.
I’d like it to Self Detect on any line.

I also am not sure what or how I can connect a Network Only Printer/Scanner/Fax Machine at home to it. It has a Telephone Jack but I’m only using VOIP.

I’m pretty advanced with Asterisk now and can figure things out..I would just like some advice and direction before I get started.

Oh, one more thing. Is there any way to Route the Faxes to different folders (extensions) because I have End Users with Phone + Extensions when you call in.