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Phone Records Lookup

Phone records lookup is a policy you might want to apply in your company in order to get sure that your resources are being used in a proper way, by analyzing where the calls being made. Or maybe the scenario is that you have developed an application for Asterisk to accept creditcards using a payment gateway and you need to look for a reverse phone lookup services in order to get the full address of the caller (not just state,city).

Whatever the reason you might have to use this kind of services, here we provide some options that you should consider some provider that might even provide APIs, so the integration of forms might be possible.

New Releases for Asterisk Are Now Available

Recently The Asterisk Development Team announced the release of Asterisk versions 11.0.2, 10.10.1 and 1.8.18.1 and made them available for immediate download at:

All of the releases resolves one or more issues reported by the community, without whose participation it wouldn’t have been possible.

The following is the issue resolved in this release:

  •  chan_local: Fix local_pvt ref leak in local_devicestate(). (Closes issue ASTERISK-20769. Reported by rmudgett)

Please read the change logs for a full list of changes. Thank you for your continued support of Asterisk!

What Is A Telecommunication System

As its name implies, a telecommunication system is first a communication system, that is, a system to transmit information from one location to another. And secondly, a system that allows you to transmit information from locations with a considerable distance between them. That’s what the prefix tele means in greek = distant.

As in any communication system, it’s composed by 3 parts:

  • The transmitter
  • The channel
  • The receiver

and it can be analog and digital.

The Components of a Communication System

Basically, a communication system is composed by three parts: The transmitter is the first component. That’s the part of the communication system that sits at point A. It includes two items: the source of the information, and the technology that sends the information out over the channel. The second component is the channel. The channel is the medium (the conduct) that the information travels through in going from point A to point B. An example of a channel is optical fiber, or the air. Finally, there’s the receiver, the part of the communication system that sits at point B and gets all the information that the transmitter sends over the channel.

An example of communication system would be a conversation between you and one friend. The transmitter would be the person talking (and the two items would be the vocal cords and the windpipe). The channel in this example would be the air between persons speaking, and the Receiver would be the other person that is listening while the first one speaks.

Scheduled Maintenance for Asterisk Project community services

On Friday, November 30th, 2012, the Asterisk community services listed below will be undergoing maintenance (migration to a new server and software upgrades). The services will be shut down at approximately 10:30 AM CST (4:30 PM December 1st UTC), and should return no later than 11:30 AM CST. Please keep in mind that it may take longer for our DNS updatesto propagate throughout the Internet. We apologize in advance for any inconvenience this may cause.


The affected services are:

git.asterisk.org

Asterisk 11.0.0 Now Available!

The Asterisk Development Team is pleased to announce the release of Asterisk 11.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

Asterisk 11 is the next major release series of Asterisk. It is a Long Term Support (LTS) release, similar to Asterisk 1.8. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 11, please see the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11

A short list of new features includes:

  • A new channel driver named chan_motif has been added which provides support for Google Talk and Jingle in a single channel driver. This new channel driver includes support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk, hold, unhold, and ringing notification. It is also compliant with the current Jingle specification, current Google Jingle
  • specification, and the original Google Talk protocol.
  • Support for the WebSocket transport for chan_sip.
  • SIP peers can now be configured to support negotiation of ICE candidates.
  • The app_page application now no longer depends on DAHDI or app_meetme. It has been re-architected to use app_confbridge internally.
  • Hangup handlers can be attached to channels using the CHANNEL() function. Hangup handlers will run when the channel is hung up similar to the h extension; however, unlike an h extension, a hangup handler is associated with the actual channel and will execute anytime that channel is hung up, regardless of where it is in the dialplan.
  • Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial allows you to execute a dialplan subroutine on a channel before a call is placed but after the application performing a dial action is invoked. This means that the handlers are executed after the creation of the callee channels, but before any actions have been taken to actually dial the callee channels.
  • Log messages can now be easily associated with a certain call by looking at a new unique identifier, “Call Id”. Call ids are attached to log messages for just about any case where it can be determined that the message is related to a particular call.
  • Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in Asterisk. Unlike traditional ACLs defined in specific module configuration files, Named ACLs can be shared across multiple modules.
  • The Hangup Cause family of functions and dialplan applications allow for inspection of the hangup cause codes for each channel involved in a call. This allows a dialplan writer to determine, for each channel, who hung up and for what reason(s).
  • Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() lets you set some of the configuration options from the general section of features.conf on a per-channel basis. FEATUREMAP() lets you customize the key sequence used to activate built-in features, such as blindxfer, and automon.
  • Support for DTLS-SRTP in chan_sip.
  • Support for named pickupgroups/callgroups, allowing any number of pickupgroups and callgroups to be defined for several channel drivers.
  • IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation

A full list of all new features can also be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/11/CHANGES

For a full list of changes in the current release, please see the ChangeLog. http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0

Thank you for your continued support of Asterisk!