Fluke launches new network tool for VoIP monitoring
It measures quality of experience for every VoIP call at every site throughout a network, in addition gauging proper network configuration.
Dev: Test my test-branch!
The developer team for Asterisk not only consists of coders - a very important part are the testers, those that test new code and give feedback.
For a few weeks, I've been maintaining a large number of branches with various stuff in them and have gotten very little feedback, not enough to judge whether or not to move forward with these patches. Some, but not all, code is written by me. There are large contributions from other developers, code that I maintain in several open subversion branches in order to help them stay up to date with their work.
To assist the testing group and make life easier, I've combined a lot of patches into one superbranch for testing. I've added the README further down.
** PLEASE help the community, please test this branch.
Check it out like this:
SVN Checkout
Then cd into test-trunk and run "make" then "make install"
Report any bugs in the proper open bug in the bug tracker. If you like new functions, add a comment that this works for you. Provide feedback, make our work easier.
Run "svn update" from time to time to get the latest version. Any changes from trunk will be merged into this code. Read the README.test-this-branch file to get more information.
Thank you for your help!
P.S: Obviously, this is test code, not recommended to be closer than 2 miles from your production servers.
----- README.test-this-branch
----------
TESTING BRANCH - WELCOME!!
----------
Asterisk is developed by the Asterisk.org user community. The development team does not only consist of coders, but also testers and people that write documentation and check for security problems.
This is a combined branch of many patches and branches from the bug tracker that needs your testing. Please test and report your results in the bug tracker reports for each patch.
What's in this branch?
----------------------
This branch includes the following branches:
- sipdiversion: Additional support for the Diversion: header
- jitterbuffer: Jitterbuffer for RTP in chan_sip (#3854)
- videosupport: Improved support for video (#5427)
- peermatch: New peer matching algorithm (no bug report yet)
- rtcp: Improved support for RTCP (#2863)
- dialplan-ami-events: Report dialplan reload in manager (#5741)
- sipregister: A new registration architecture (#5834)
- subscribemwi: Support for SIP subscription of MWI notification (#6390)
Coming Here Soon:
- iptos: New IPtos support, separate audio and signalling (#6355)
- metermaids: Subscription support for parking lots (#5779)
- multiparking: Multiple parking lots (#6113)
And the following stand-alone patches
- New CLI commands for global variables (#6506)
- Additional options for the CHANNEL dialplan function
All of these exist in the bug tracker
* PEERMATCH: New object match for incoming calls. Skip the "user" :-)
---------------------------------------------------------------------
In this code, we will match incoming calls like this:
- First user on From: user name
- Then peer on From: user name *** NEW ****
- Then peer on IP and port
This means that in most configurations, you can configure a phone entry as "type=peer" instead of "type=friend". Subscriptions will work much better with just one object to match.
/Olle
For a few weeks, I've been maintaining a large number of branches with various stuff in them and have gotten very little feedback, not enough to judge whether or not to move forward with these patches. Some, but not all, code is written by me. There are large contributions from other developers, code that I maintain in several open subversion branches in order to help them stay up to date with their work.
To assist the testing group and make life easier, I've combined a lot of patches into one superbranch for testing. I've added the README further down.
** PLEASE help the community, please test this branch.
Check it out like this:
SVN Checkout
Then cd into test-trunk and run "make" then "make install"
Report any bugs in the proper open bug in the bug tracker. If you like new functions, add a comment that this works for you. Provide feedback, make our work easier.
Run "svn update" from time to time to get the latest version. Any changes from trunk will be merged into this code. Read the README.test-this-branch file to get more information.
Thank you for your help!
P.S: Obviously, this is test code, not recommended to be closer than 2 miles from your production servers.
----- README.test-this-branch
----------
TESTING BRANCH - WELCOME!!
----------
Asterisk is developed by the Asterisk.org user community. The development team does not only consist of coders, but also testers and people that write documentation and check for security problems.
This is a combined branch of many patches and branches from the bug tracker that needs your testing. Please test and report your results in the bug tracker reports for each patch.
What's in this branch?
----------------------
This branch includes the following branches:
- sipdiversion: Additional support for the Diversion: header
- jitterbuffer: Jitterbuffer for RTP in chan_sip (#3854)
- videosupport: Improved support for video (#5427)
- peermatch: New peer matching algorithm (no bug report yet)
- rtcp: Improved support for RTCP (#2863)
- dialplan-ami-events: Report dialplan reload in manager (#5741)
- sipregister: A new registration architecture (#5834)
- subscribemwi: Support for SIP subscription of MWI notification (#6390)
Coming Here Soon:
- iptos: New IPtos support, separate audio and signalling (#6355)
- metermaids: Subscription support for parking lots (#5779)
- multiparking: Multiple parking lots (#6113)
And the following stand-alone patches
- New CLI commands for global variables (#6506)
- Additional options for the CHANNEL dialplan function
All of these exist in the bug tracker
* PEERMATCH: New object match for incoming calls. Skip the "user" :-)
---------------------------------------------------------------------
In this code, we will match incoming calls like this:
- First user on From: user name
- Then peer on From: user name *** NEW ****
- Then peer on IP and port
This means that in most configurations, you can configure a phone entry as "type=peer" instead of "type=friend". Subscriptions will work much better with just one object to match.
/Olle
Cisco IP Contact Center Solutions to be Deployed by Cox Communications
SAN JOSE, Calif., February 21, 2006 - Cisco Systems, Inc. today announced that Cox Communications will standardize on Cisco Internet Protocol Contact Center (IPCC) solutions to provide customer service and support to its subscribers. By implementing Cisco’s standards-based IP software, Cox plans to transform business processes by streamlining management of its distributed contact centers and reducing operational costs. Cisco named Cox as its 3000th IP Contact Center customer today. < …
Introducing Telephone Reminders 2.5: The Asterisk Telephone Reminder System
Excerpt: Using nothing but a phone call, you can schedule reminders for the near or distant future, specify different numbers for the return calls, and customize a recorded message for each call. In short, it's perfect for appointment reminders, birthday reminders, anniversary reminders, and anything else you want or need to remember.
Click Here for the Full Nerd
Click Here for the Full Nerd
Create your own Voice-over-IP PBX using Asterisk
Asterisk is already hard at work in South Africa. Its being used as a PABX, for call-recording, for both small and large call-centres, for voice conferencing, and in CTI (Computer Telephony Integration). Its providing inter-office "free" calling, and very inexpensive international calling. In every case, Asterisk-based solutions are a fraction of the cost of the traditional equivalents.
Voice-over-IP (VOIP) is built right in to Asterisk. Connecting into the traditional phone network is done using interface cards readily available in South Africa, or by connecting into an ITSP (Internet Telephony Service Provider) via VoIP over the Internet.
Asterisk provides all the functions of even the most expensive traditional PABXes simply with software running on an ordinary PC. What's more it comes with all the open-source goodness that Tectonic readers know and love.
At Connection Telecom we like to say that the coming together of VoIP and the open source world is resulting in the most dramatic change in the world of telephony since we last heard "Nommer Asseblief?" ["Number Please" in Afrikaans].
Click Here for the Full Article
Voice-over-IP (VOIP) is built right in to Asterisk. Connecting into the traditional phone network is done using interface cards readily available in South Africa, or by connecting into an ITSP (Internet Telephony Service Provider) via VoIP over the Internet.
Asterisk provides all the functions of even the most expensive traditional PABXes simply with software running on an ordinary PC. What's more it comes with all the open-source goodness that Tectonic readers know and love.
At Connection Telecom we like to say that the coming together of VoIP and the open source world is resulting in the most dramatic change in the world of telephony since we last heard "Nommer Asseblief?" ["Number Please" in Afrikaans].
Click Here for the Full Article
More VOIP News…
View more VOIP news and analysis from Computerworld.com.
Asterisk on OpenWrt
Asterisk is free software that lets you create a fully functional, easily customizable, private branch exchange (PBX). Businesses like Asterisk because they can save money by using it, and because it is open source, they can add functionality to it easily and inexpensively. Asterisk is also becoming popular with home office users -- so much so that it spawned a new project called Asterisk@Home,
Voice over IP in the European Home: a Discussion with Patrick Lelorieux of Linksys
February 17, 2006 Last year Linksys won the best voice-over-IP (VoIP) product category in the Wireless Broadband Innovation Awards for its Wireless-G Router. The accolade represented much more than a worthy addition to Linksys’ prize cabinet; VoIP is arguably one of the hottest topics in home …
What if they had Skype on 'Lost'?
those of you who don’t watch the television show ‘Lost’, this may be a real stretch, but I’ll try to reel it in for you
anyway. There’s a scene where a guy named Michael, whose son Walt has been kidnapped, sits down in front of an Apple IIe
desktop (yeah–the ancient 8 bit monster) and accidentally discovers that he can chat with Walt through the computer
using some simple kind of instant messaging software that appears to be "pre loaded" on the computer.
Anyway, Michael is desperate to rescue his son, and is therefore very excited to discover that his son is trying to
communicate with him using the PC. When this particular episode aired, maybe a week or two ago, I remember
thinking to myself, "too bad that Apple II can’t run Skype". I know, I know. VoIP pervades
so many aspects of my life. But anyway. Here’s a
about. Perhaps the Dharma Initiative will consider issuing more VoIP-capable PCs when it decides to do its next human
captivity experiment on a deserted island in the middle of the ocean.
Article: Open source Asterisk PBX getting more Popular
Asterisk, the open source PBX system made by Digium, is gaining ground with companies and governments alike. The cost savings over proprietary PBX systems can be substantial, but Mark Spencer, president of Digium, told TG Daily in a short interview at the Southern California Linux Expo, that "choice" is the main reason companies adopt Asterisk. "Customers have a choice in how they configure they system, in the hardware they buy or the graphical interface they want," says Spencer. This choice also allows companies to add extensions or make changes in seconds compared to days with a traditional PBX.
Click Here for the Full Article
Click Here for the Full Article