Asterisk 10 Beta 1 From New Asterisk 10 Branch Is Ready For Downloading And Testing

http://snapvoip.blogspot.com/
The Asterisk Development Team announced the availability of the first beta release of Asterisk10, Asterisk 10.0.0-beta1. The release is available download and consumption at Asterisk download site.

As we mentioned before, Digium and Asterisk.org has dropped "1." from the Asterisk version numbers and the new Asterisk 10 branch, will continue to march forward.
Of course Asterisk team is requesting your participation in testing, we are testing it with PIAF-RED;

All interested users of Asterisk are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk. Additionally users can make use of the RPM and DEB packages now being built for all Asterisk releases. More information

Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk versions page:

A short list of included features includes:

* T.38 gateway functionality has been added to res_fax.
* Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
* New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
* Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application.
* Support for defining hints has been added to pbx_lua.
* Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
* Much, much more!

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

http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog

OpenSIPS 1.6 eBootcamp, Get Ready For A Torrent Of SIP Technology

OpenSIPS 1.6 eBootcamp http://snapvoip.blogspot.com/

OpenSIPS is holding a eBootcamp on OpenSIPS 1.6. This bootcamp is unique in itself because it is held over the Internet for over 7 weeks. Participants will learn from downloading installing and configuring and on to administration of OpenSIPS. It will start on September 19th 2011. The ebootcamp will accept late registrations up until 12th September 2011. So you better hurry. Contact bootcamp@opensips.org for registration and more information

The users will learn how to authenticate users, install a GUI to help with daily administration, forward calls to the PSTN through Dialplan, integrate Asterisk and Voice Mail, Presence agent, Load Balancing, Traverse Nat for SIP and generate CDR records to a Radius Server. At the end, you will learn how to use troubleshooting tools to solve end user problems.

This is basically the same OpenSIPS Bootcamp but help over a broadband connection over a longer period, seven weeks to be exact.

"The live classes will be taken online by web-conference every Tuesday and Thursday 03:00PM GMT, 11:00AM EDT 08:00AM, PDT. To attend this training you will need to have broadband Internet access. You are going to receive a DVD with a virtual machine to run the labs. The virtual machine will be available in the VMWARE format. You can download the free VMWARE player to run the VM. We suggest that you have one separate desktop or server for your VM and at least one IP Phone/ATA in your private labs to complete the training. A LMS (Learning management system) will be available with forums, quizzes and support materials."

The course of learning will cover key objectives;

Install OpenSIPS on a Linux Machine

Routing basics and the default configuration

OpenSIPS authentication using MySQL and Memcache

Install OpenSIPS control Panel.

Connect to the PSTN using Dialplan and Dynamic Routing

Voicemail integration using Call Forward and AVPs

Implement a presence agent

Understand important aspects of load balancing and high availability

Implement SIP NAT traversal using RTPProxy

Account Calls to MySQL

How to use test and monitoring tools to check your configuration

OpenSIPS

SIP Forum Releases SIPit Interoperability Test Results from SIPit 28 Held In in Huntsville, Alabama

SIPit 28 SIP Interoperability http://snapvoip.blogspot.com/

SIP Forum has released the test results from the latest SIPit event SIPit 28, the SIP interoperability event, was hosted by Digium on April 11–15, 2011 and held at the Jackson Center in Huntsville, Alabama's Research Park. SIPit is organized by the SIP Forum's Test Event Working Group (TEWG) and serves as a "plugfest" for participating companies to perform SIP interoperability testing with other participants in a live network environment. There were 19 companies from 10 countries who participated in the SIPit 28 to test and confirm SIP interoperability this spring. High definition IP video was a key focus at the event and IPv6 support in the industry was also on the rise, the forum noted.

"SIPit28 was one of the smoothest events to date featuring a lot of positive energy and hard work from all involved, We received great feedback from participating companies who were able to glean a good deal of valuable data which provided them a very high return.", said Robert Sparks, chair of the SIP Forum's Test Event Working Group and organizer of SIPit 28.said Robert Sparks, chair of the SIP Forum's Test Event Working Group and organizer of SIPit 28.

Follow the links after the jump.

Following is the press release by SIP Forum.

NORTH ANDOVER, MA (June 27, 2011) - The SIP Forum has announced the results of its latest SIP Interoperability Testing event, SIPit 28, the world's premier interoperability testing event for the Session Initiation Protocol (SIP) for application developers, service providers, and IP communications equipment manufacturers, held earlier this spring at Jackson Center in Huntsville, Alabama's Research Park. Hosted by Digium, the five-day testing event once again demonstrated the critical role that Session Initiation Protocol (SIP) plays in today's telecommunications industry while also highlighting the growth, evolution and adoption of SIP services in fixed and mobile IP networks worldwide.

SIPit 28 provided 19 attending companies with valuable "real-world" deployment data gathered from the 40 distinct SIP implementations conducted during the week as participants tested SIP interworking in a variety of simulated, live IP network conditions and settings. Testing exercised the full range of defined transport protocols for SIP, from UDP to TLS over SCTP.

As part of the week-long testing event, which took place April 11-15, participants showed a growing interest in deployment of high definition (HD) video services across global IP networks. Participants executed a number of different telepresence scenarios with intricate SIP signaling, testing for quality of service (QoS), network resiliency and interoperability involving new types of rich media endpoints with IP application servers, SIP proxy servers and session border controllers.

"Rolling out IP video in both enterprise and service provider networks and making it work is a key objective for many operators, equipment makers and application developers in today's telecom world and is something attending companies really focused on at this year's SIPit" said Robert Sparks, chair of the SIP Forum's Test Event Working Group and organizer of SIPit 28.

"We were able to successfully execute a number of testing scenarios that examined different network topologies under a variety of simulated conditions. Participants were able to verify correct behavior of advanced applications in an environment that is very difficult to duplicate without so many different implementations in the same room at the same time."

In addition, IPv6 was also a hot topic in preparation for the industry-wide rollout of this next generation Internet protocol, with many participants uncovering new, positive information surrounding interoperability of IPv6 equipment and services, particularly when implementations use a combination of IPv6 and IPv4 for media and signaling. Across the testing, 68 percent of the implementations at the event supported IPv6 which represented a significant increase from SIPit 27 in 2010 which saw 53 percent of attendees achieve IPv6 support.

A full summary report of SIPit 28 results is available at the SIP Forum website or by clicking here (https://www.sipit.net/SIPit28_summary).

"The SIPit events continue to be critically important and popular events for the IP communications industry because they give participants an opportunity to vigorously test implementations of their own IP communications product or service with other members of the telecom community in a non-commercial, real-world testing environment," said Marc Robins, SIP Forum President and Managing Director. "Every time we have a SIPit event, we receive scores of positive feedback about how valuable the testing was to the development of products and services, as well as how useful it is to standards bodies such as the IETF for gaining insight into the growth of SIP and the specific SIP implementations."

SIPit is organized by the SIP Forum's Test Event Working Group (TEWG) and serves as a "plugfest" for participating companies to perform SIP interoperability testing with other participants in a live network environment. Conducted twice a year, with events rotating in the United States, Europe, and Asia, the SIP Forum has hosted 28 events around the globe. In the coming months, the SIP Forum will release details about its next event - SIPit 29 - which will be hosted by ETSI and take place this fall in France. For information about past SIPits, please visit www.sipit.net.

"SIPit28 was one of the smoothest events to date featuring a lot of positive energy and hard work from all involved," continued Sparks. "We received great feedback from participating companies who were able to glean a good deal of valuable data which provided them a very high return."

SIPit via SIP Forum

SIP/IP Trunking Market Report & Forecast 2010 For UK

http://snapvoip.blogspot.com/
Research & Markets has published “UK IP/SIP Trunking Market Report & Forecast 2011”, a report and forecast on SIP, SIP trunking etc. SIP Trunking is experiencing a sharp growth in UK. The 20 page report carries a wealth of information to interested parties. Below is the table of contents and the press release;

1 Executive Summary

2 Market Definitions

3 Market Forecast

4 Market Dynamics
4.1 Strategies
4.1.1 Hosted Enablement
4.1.2 PBX Manufacture teaming
4.1.3 ISDN Replacement
4.1.4 Cloud Communications/Computing
4.1.5 Business Continuity
4.1.6 PBX Enhancement
4.2 Vendor Propositions
4.2.1 The Changing Proposition
4.2.2 IP-PBX Interoperability
4.3 Relationship to PSTN/ISDN dynamics
4.4 Pricing
4.4.1 Pricing Comparisons

5 UK SIP Trunking Service Providers

6 Challenges to Growth

7 Technology Factors
7.1 Access & QoS
7.2 Gateways
7.2.1 Interop with ITSPs
7.2.2 Survivability
7.2.3 The Future of Gateways
7.2.4 Gateway Vendors

DUBLIN--(BUSINESS WIRE)--Research and Markets (http://www.researchandmarkets.com/research/658681/uk_ipsip_trunking) has announced the addition of the "UK IP/SIP Trunking Market Report & Forecast 2011" report to their offering.

“UK IP/SIP Trunking Market Report & Forecast 2011”

As the market for IP trunking breaks 231,000, and the UK trunking market enters a period of sharp growth. illume launches its 2011 UK IP/SIP Trunking Market report and forecast. As usual illume provides a unique insight into the market, based on its regular market interviews and quarterly industry survey. The report is authored by Matthew Townend and highlights include:

• Market Forecast (2010 - 2013)
• Market Dynamics (Strategies, Vendor Propositions and Pricing)
• Identification of the UK's SIP Trunking Providers
• Challenges to growth
• Technology factors (QoS, Gateways, Broadband etc)

This 20 page report provides insight into the current market numbers and likely growth over the next 3 years. illume has also undertaken a detailed review of the current service provider propositions and strategies. As customers and providers look at SIP Trunking as a possible recession proof service the purchase of this report will enable its readers to gain a better understanding of the market opportunity that really exists.

To purchase the report click the "Buy Now button" and proceed to check out where you can select your preferred method of payment. After payment has been made you will be emailed your UK IP/SIP Trunking Market Report & Forecast 2011 as a PDF file within 24 hours.

For more information visit http://www.researchandmarkets.com/research/658681/uk_ipsip_trunking

Contacts

Research and Markets
Laura Wood, Senior Manager
press@researchandmarkets.com
U.S. Fax: 646-607-1907
Fax (outside U.S.): +353-1-481-1716

The Secrets of SIP Trunking : A Seminar At IAUG

The Secrets of SIP Trunking http://snapvoip.blogspot.com/
This Seminar aims to show people how SIP trunking is displacing traditional technologies on its way to becoming the leading service for Voice provisioning. The focus is in how SIP trunks works along with ensuring they are Secure, Reliable and able to deliver well into the future and offer new features as they become available. The Seminar also fully intends to shake things up and tell people what to watch out for when entering the SIP trunk market and why things will not always go as planned. With a whole list of Gotchas to run through, delegates will learn that SIP trunking is most certainly the future but the road to this end will not be a smooth one.By meeting the Seminar objectives of Informing, guiding and making people think; delegates will walk away with an understanding of SIP Trunking along with a lot of food for thought on a technology that's new and exciting but can also be fraught with interoperability issues that can quite often makes grown men cry.

Session Number: 10024
Track: Professional Development
Room: Neopolitan 4
Session Type: General Sessions
Presenter: (The SIP School)
Time: May 25, 2011 (08:30 AM - 09:30 AM)

REGISTRATION SITE IS HERE 

Professional Roadmap: Managerial
Experience Level: Basic/Introductory Overview
Category: General Overview
Objective #1:: Generate an understanding that SIP Trunking is the Future and to prepare for it.
Objective #2:: Understand how to approach Service providers in order to understand what they do offer and should offer an enterprise. i.e. How not to get caught out with a 'poor' provider.
Objective #3:: Understand some of the technical challenges of getting SIP Trunking right 'before' you even talk to a provider.

Keep Your PBX in a Flash ( PIAF ) Running During An Internet Outage!

PBX in a Flash ( PIAF ) Can Run without Internet http://snapvoip.blogspot.com/
If you experienced PIAF server was not able to route any calls after losing your Internet connection, even between local extensions Ward Mundy just Tweeted how we could avoid this.
The solution is to run a Bind DNS Caching Server running on your PIAF box to keep it going with an Internet outage. If you decided to use this solution, and if you are using FDQNs on your phones, the primary DNS server address must internal or external IP address of your server, where your phones are attached.
Ward Mundy, on the PIAF forum, has provided a simple script to run on your server and a set of files to get everything setup properly like ROOT server information and forwarding DNS queries to Google DNS servers.. The script is not shell script so you will have to run each line by themselves. (You can convert it into a shell script, if you have many PIAF servers running.)
After logging into your server with root capability (root or a sudo account) run the following code at the end of the article. Once you run the code and satisfied that everything is fine, run a dig command (e.g. dig inbound1.vitelity.net) for each of the FQDNs of your SIP trunk providers! This will cache the DNS information on your server to be used when the need arises. You can find the info with FreePBX: Tools, Asterisk Info, SIP Info, SIP Registry. Happy calling;

yum -y install bind*

cd /var/named
service named stop
wget http://pbxinaflash.net/source/bind/bind.tgz
tar zxvf bind.tgz
rm bind.tgz

echo nameserver 127.0.0.1 > /etc/resolv.conf

sed -i 's|$nameserver|127.0.0.1|' /sbin/dhclient-script

service named start
service network restart

dig pbxinaflash.com
dig pbxinaflash.com

# look at the output for the second dig command issued above
# 2d dig command should show a very short DNS lookup time, e.g.
#;; Query time: 1 msec
#;; SERVER: 127.0.0.1#53(127.0.0.1)
# IF ALL IS WELL...

#be sure to set named for auto start on bootup

chkconfig  named  on --level 2345
chkconfig --list named

# named should be configured to start with RunLevels 2,3,4,5

SIP Network Operators Conference ( SIPNOC ) Preliminary List Of Registered Attendees And Participants Announced.

SIPNOC 2011http://snapvoip.blogspot.com/
SIPNOC seem to be going places with ever expanding attention from CEOs, CTOs, network architects and engineers from carriers on five continents as well as government, research organizations and industry bigwigs. The first of its kind conference will be a good source of information for everyone involved in SIP Network operations or any communications operations involving SIP for that matter.
SIPNOC earlier announced that the Keynote Speech will presented by Dr. Douglas Sicker, FCC.

"When we initiated SIPNOC, we knew we were embarking on a new type of event for the industry and that there would be interest, and the response we have received has justified our enthusiasm, and then some, We expect this event to be far more than just an educational conference, but a meeting of the brightest industry minds from across the entire spectrum of the global networking operator community. Our vision for SIPNOC is to make it a place for network operators of all shapes and sizes to come together to share their perspectives and ideas on how to make IP communications better, and hopefully provide tangible results that further accelerate the development and propagation of SIP and SIP services in the years ahead." " said SIP Forum Managing Director, Marc Robins.

Follow the links below for registration and further information.

Press Release;

NORTH ANDOVER, MA (March 18, 2011) - The SIP Forum announced today a preliminary list of registered attendees and participants in the first SIP Network Operators Conference (SIPNOC), to be held April 25-27 at the Hyatt Dulles Hotel in Herndon, VA. The two-day conference, which is designed to be a first-of-its-kind international conference aimed a SIP network operations personnel, is attracting CEOs, CTOs, network architects and engineers from carriers on five continents, including AT&T, Comcast, Cbeyond, Time Warner Cable, Swisscom, COX Communications, Verizon, Uni-tel, A1Telekom Austria, Telecommunikasi Indonesia, XO Communications, and Sprint Nextel.

In addition to carrier participants, SIPNOC has also attracted a myriad of SIP community stakeholders from vendors, governments and research organizations such as Acme Packet (which has also signed on as a SIPNOC Diamond Sponsor), the FCC, Avaya, Cisco Systems, Microsoft Corp., IBM Research, CableLabs, Dialogic, Broadsoft, Ingate Systems, MetaSwitch, Polycom, Commetrex, Sonus, Sagemcom Canada, NGN Data Services Corp. and the Illinois Institute of Technology.

Registration for SIPNOC remains opens by visiting http://www.regonline.com/sipnoc_2011.

"When we initiated SIPNOC, we knew we were embarking on a new type of event for the industry and that there would be interest, and the response we have received has justified our enthusiasm, and then some," said SIP Forum Managing Director, Marc Robins. "We expect this event to be far more than just an educational conference, but a meeting of the brightest industry minds from across the entire spectrum of the global networking operator community. Our vision for SIPNOC is to make it a place for network operators of all shapes and sizes to come together to share their perspectives and ideas on how to make IP communications better, and hopefully provide tangible results that further accelerate the development and propagation of SIP and SIP services in the years ahead."

The SIP Forum has also developed a world-class lineup of speakers at SIPNOC, to be announced in detail at a later date.

SIPNOC will focus on issues critical to the reliable and successful deployment and operation of SIP-based services in carrier networks. The agenda will feature special presentations, panel discussions and workshops covering key topics by network operators related to SIP-based services and infrastructure, including testing, application development, SIP trunking, FoIP, call routing and peering, troubleshooting and monitoring, emergency services and more.

While the international carrier and service provider community is the lynchpin of the event, industry stakeholders involved in major SIP initiatives such equipment vendors, government representatives, large enterprises and research organizations are encouraged to attend.

The SIP Forum has gained an international reputation for developing important, educational events surrounding SIP. The organization's SIPit series of interoperability testing events (www.sipit.net) regularly provides a test bed for SIP-based applications and equipment that has been heralded as critical for the development of new products and services in the industry. The SIP Forum also has a number of committees and task groups made up of well-known industry experts examining a myriad of SIP-related topics, including the use of SIP in smart grid installations, FoIP, video, and user-agent configuration.

SIPNOC Registration

* To register for SIPNOC, please visit http://www.regonline.com/sipnoc_2011.

SIPNOC General Show Info

* For more information about SIPNOC, please visit www.sipnoc.org or send an email to sipnocinfo@sipforum.org.

SIPNOC Sponsorship Info

* For information about corporate sponsorship opportunities at SIPNOC, please contact Marc Robins, SIP Forum President and Managing Director, at +1-718-548-7245 or marc.robins "at" sipforum.org.