If after installing Asterisk 10.X.X you see the following errors on SIP reload: No valid Transports Available, Falling Back To UDP, This might be because a possible regression. Please open an issue in JIRA referencing th..
Author : Pavel Espinal
Home » Articles posted by Pavel Espinal (paged 6)
Due to high volume of outbound calls you might be asked to alternate the outbound traffic by your termination provider.Basically, what what you need to do here is: Having 2 destinations (Dst_A, Dst_B), check/set a variable in the AstDB. Then if the varia..
We are currently looking for a PHP/MySQL senior developer in the Dominican Republic. We are offering a full contract with excellent salary.Requirements: OOP PHP MySQL knowledge (you dont have to be a ninja but you must have a good level) jQuery M..
If your Asterisk box is facing the Internet and you are being constantly flooded with break-in attempts, you might want to follow these steps in order to secure your Aster..
The news of Digium IP Phones have got a big number of people giving their opinion on the subject. Some of the inquiries are about the manufacturing process, others even question the very reasons for Digium to launch phones (after all, they are enter..
The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/The release of Asterisk 1.8.9.0 resolves several issues repor..
Asterisk cannot act as a proxy, it is a B2BUA. If you want to make its behavior appear to be a proxy, there are a number of things you can do, but it will never just ‘pass along’ headers from an incoming INVITE to an outgoing INVITE.When Aster..
Is it possible to make Asterisk jump into action and play a sound file as soon as a handset is lifted, instead of providing a Dial Tone and waiting for the user to dial an extension.With analog phones (chan_dahdi) – you just have to set ‘immedi..
AstLinux is a custom Linux distribution centered around Asterisk, the Open Source PBX. Although Asterisk remains the core focus, other VoIP applications such as FreeSWITCH are also available. AstLinux has many unique features that make it ideal for embed..
H.323 Video Conferencing with GNU Gatekeeper has been improved with the new released announced by Jan Willamowius, Founder of the GNU Gatekeeper Project.A new major release of the GNu Gatekeeper has been anounced. This new version has many new featu..