PaloSanto Solutions silver sponsor at Asterisk World

PaloSanto Solutions silver sponsor at Asterisk World

Elastix and PaloSanto Solutions will be present at the next ITExpo East on the 2nd of February, 2011.

ITExpo is part of a dual event that TMC develops in Miami and Los Angeles, at the beginning and midyear, respectively. ITExpo gathers the most representative exponents of the telecommunication industry with keynote conferences and product exhibition.

Elastix will be present and you will have the opportunity to visit our stand and talk with important members of the Project and Gold Resellers of the region. See exhibitor list.

As part of our participation we will make the launch of some products, enlist the certified hardware available at the moment and will announce the 2010 community awards winners.

Elastix is also Silver Sponsor for Asterisk World, a collocated event inside ITExpo and dedicated to this important telephony distribution. We will present a keynote there the 3rd of February at 12h30 Miami.

http://www.tmcnet.com/voip/conference/digium-asterisk-world/east-11/

Join us at this event, we will be at Booth #1009. For a free pass to visit our booth you can register here

We hope to see you all

Sincerely

 

"Elastix is a stable, reliable and flexible system, built to serve both: integrators and end users. For these reasons we chose it as the default operating system in our IP-PBX solutions, and market feedback confirms this was a good decision."

Eran Gal - CEO

Xorcom Ltd

Promotion of VoIP Business Telephone Solutions by Toshiba America Information Systems

As it is described in the last press release, Toshiba America Information Systems Inc., Telecommunication System Division, and LEAF Financial Corporation (LEAF), the commercial finance subsidiary of Resource America Inc. have initiated a vendor finance program agreement to foster the customer financing and leasing requirements of the Authorized Toshiba Dealer Network.

The Program will be named as Toshiba Telecom Finance, in order to be the tool for authorized Toshiba Dealers to provide comprehensive lease finance offering, targeted to make Toshiba VoIP business telephone solutions more available to their small to medium business customer pool. "We are very excited to work with LEAF Financial to provide our customers with flexible financing solutions from an industry leader in the small and middle-market telephony leasing space,” said Larry Meyer, vice president of Sales and Marketing of Toshiba America Information Systems for the Telecommunication Systems Division. “We are committed to providing the Authorized Toshiba Dealer Network with innovative products and programs that will assist them in maximizing their telecom sales efforts." "We are pleased that Toshiba selected LEAF to provide financial services to Authorized Toshiba Dealers and their customers and to support Toshiba Telecom Finance," said Crit DeMent, CEO of LEAF, in a press release. "As companies invest in leading-edge communication solutions to stay competitive, Toshiba Telecom Finance will help them achieve those goals with affordable financing solutions. LEAF's highly responsive service and support capabilities will make it extremely convenient for the Authorized Toshiba Dealer Network to utilize lease financing as a powerful closing tool." LEAF Financial Corporation is headquartered in Philadelphia, Penn., being the commercial finance and asset management company. Its business model is designed to emit small to middle sized equipment leases and loans in a variety of asset ranging through specialized strategic business units. After origination of the leases, LEAF operates these leases for its own account, institutions, and individual investors by means of investment partnerships and other different forms of investment.

Free BT device could produce faster VoIP solutions

A device being offered for free to customers of major internet service provider BT could help people use VoIP hardware more efficiently, reports ISPreview.co.uk. The BT Broadband Accelerator - formerly known as the I-Plate - is a simple device that can be plugged in to domestic and business phone lines and reduces local electronic interference to produce faster broadband connection speeds.

Existing BT customers will only have to pay a small postage fee to get hold of the device, while non-BT broadband users will be charged around £8 for the product itself.

Broadband connection speeds can be affected by electronic interference generated by nearby devices such as stereos, televisions or other computers, but BT claims its new product can boost speeds by as much as 1.5 megabits per second.

The company recently announced a £1.5 billion investment in developing next-generation fibre optic networks across the UK and is currently testing the technology in nearly 70 towns and cities.

By Telappliant.

Railroad Company Amtrak selects solutions from VoIP provider Empirix

World VoIP NewsRailroad Company Amtrak selects solutions from VoIP provider EmpirixRailroad Company Amtrak selects solutions from VoIP provider EmpirixRailroad Company Amtrak selects solutions from VoIP provider EmpirixEmpirix Inc., the market leader in VoIP telecommunications market, announced that company Amtrak, United States biggest passenger railroad, has chosen the Empirix® OneSight® Voice Quality Assurance solution to monitor Amtrak’s voice network to detect and prevent voice communications shortfalls before they affect routine operations. Once it is being emphasized on ensuring Amtrak’s telephony systems operate as they have to, the OneSight Voice Quality Assurance solution makes the enterprise to proactively maintain voice quality and the quality of experience (QoE) continuosly.

“In an industry where the competition is stiff—we compete with various methods of travel, land and air alike—it is imperative that we ensure a high quality customer experience” Amtrak operates on more than 500 destinations in 46 states and three Canadian provinces exceeding 21,000 miles of railroads. Commonly, more than 74,000 passengers ride an Amtrak train daily, and more than 45,000 passengers call into the Amtrak 24/7 Service Center every day. High call volume impose massively weighted importance placed on delivering high customer QoE, it is of the utmost significance that the Service Center connects caller to relevant agent in an appropriate amount of time, while also ensuring the best quality for the customer from end-to-end. Empirix’s One Sight Voice Quality Assurance solution makes tailored and thoroughly adjusted measurements and reporting on the call experience and figures out voice quality issues that other solutions cannot calculate, providing telephony managers with a dedicated, on-line view of the operating server. By means of a preset dashboard, the OneSight Voice Quality Assurance solution makes organizations to view the realistic indicators of their telephony servers, thus simplifying the further actions, aimed to sub-areas where a voice quality relayed issues may appear, therefore fixing them in a timely manner. As a result, organizations can prevent end-user frustration, thus enhancing the customer QoE. “Ninety percent of Amtrak’s diagnostic testing that finds problems between its IVR applications and back-end data systems are now fully automated by Empirix,” said Tim Moynihan, vice president of marketing, Enterprise business unit, Empirix. “Using our solution, Amtrak can update their IVRs and test them in a pre-deployment phase, allowing them to ensure the call happens as it should. With Empirix, Amtrak is able to obtain reports that illustrate server activity through the preset dashboard, helping them identify issues long before they can impact the end- user—all with a cost effective solution. By implementing our solution to ensure voice quality from end-to-end, Amtrak clearly demonstrates their commitment to providing the best customer experience.”

StarView Solutions New istarsip iPhone Application Available at App Store

StarView Solutions New istarsip iPhone Application Available at App Store StarView Solutions New istarsip iPhone Application Available at App StoreAustin, Texas-based StarView Solutions, a provider of business-ready Unified Communications as a Service “UCaaS” announced that it has launched a new native SIP/VoIP phone application, istarsip 3.0, in the Apple Store. This new application is the company’s first dual mode VoIP calling application for iPhone, iPod touch, and iPad. David Walsh, president of StarView Solutions said that istarsip is the company’s first phone app which turns a smart phone into a virtual VoIP phone and can reduce long distance charges while traveling or calling overseas. The application was developed because other similar applications were lacking some key functionality and feature issues that the company was able to incorporate.
The istarsip 3.0 phone application adds a second number to any iPhone, iPod touch, and iPad that works over 3G, WiFi or cellular networks. It is completely automated and requires no hardware or software configurations to use the application. Additionally, it can seamlessly integrate with StarView’s Hosted PBX solution as well as traditional office and home phones.
As opposed to other applications that share phone lines, every user of istarsip has a dedicated line. It also has enterprise capability which enables users to turn their mobile phone into an office extension to send and receive calls over their company’s phone system.
Available as a free download from Apple Inc.’s App Store at www.itunes.com/appstore, the new istarsip 3.0 supports Apple Inc.’s push notification service. With this, users will have the ability to receive notifications about new incoming calls even when the app is not running on their mobile phone.
Apart from having value-added services as well as linking mobile and phone lines, the new istarsip 3.0 also features many professional grade features such as good call quality, separate voicemail and notification, call waiting/call transfer, speaker phone support, multiple ringtones, mute/hold capability, phone number shortcuts, and more.
David Roberts, CEO of Teligistics, said, “This Starship application is a major step in controlling wireless costs for global companies. The app is easy to use and the rates are more favorable than most enterprise-level contract pricing schemes we have seen for enterprise and global customers. This is an app we will recommend to clients for controlling international wireless costs.”
StarView's service offerings include converged voice and data packages, Hosted PBX, SIP trunks, network monitoring, diagnosis and event management, private line and multi-site MPLS connections, fax-to-email services, toll-free numbers, and wholesale carrier termination.

By Jayashree Adkoli.

Could VoIP solutions be the future of emergency calls

VoIP solutions could be used by emergency service teams in the future, according to experts. During many natural disasters, standard call networks are often destroyed or damaged to such an extent they can not handle user demand.

IP telephony solutions are capable of providing continued service during such events through satellites, as was the case earlier this year in the aftermath of the Haiti earthquake.

Jack Deasy, civil programs director at satellite communications provider Inmarsat, claimed that VoIP solutions provide immediate contact between the victim and the emergency services.

He told US IT website gcn.com: "Voice is the primary communications channel people want in a disaster."

Mr Deasy added that trapped residents used static communication, such as e-mail and text messages, to alert the emergency teams to their location.

Around 30 per cent of Haiti's terrestrial phone network was wiped out by the tragedy, with much of the gap filled by IP satellite communications.

Telmar Network Technology’s Partnership with Tango Networks and UC Solutions

Telmar Network Technology declares a strategic partnership with Tango Networks to distribute its mobile Unified Communications solutions in Central and Latin America, and ensure spare parts management and out-of-warranty maintenance and repair to Tango's customers Worldwide. Telmar has accumulated almost 50 years experience on solutions that surpasses the life time of crucial network equipment, ensure massive cost savings and increase operational efficiencies for OEMs (Operation, Engineering Maintenance) Enterprises and Network Service Providers around the world.

As a designated channel partner, Telmar will distribute the Tango Networks' Abrazo™ mobile Unified Communications solution which is designed to integrate mobile phones with corporate Unified Communications networks consisting with Microsoft’ s Office Communications Server, Cisco Unified Communications Suite and merely any IP-PBX or TDM-PBX backed VoIP network. Moreover, Telmar will also deliver out-of-warranty repair for the Tango Abrazo mobile UC and spare parts processing to Tango's customers anywhere. "The addition of Tango Networks Abrazo mobile UC platform to our unified communications offerings will allow Telmar to provide a mobile UC solution to our enterprise and service provider customers around the world," said Steve Pickett, President and CEO of Telmar Network Technology. "We're also excited to provide Tango's customers cost-effective repair services for their warranty work, and offer a spare parts management solution that can help them reduce inventory carrying costs and total cost of ownership without reducing their enterprise network's reliability or performance." So called ‘ ’ Abrazo solution’ ’ from Tango Networks makes any type mobile phone to be a real extension on the corporate Unified Communications network. For the enterprise, real integration with the corporate Unified Communications system provides increased workforce productivity, outreach and customer satisfaction as a final outcome. Tango Abrazo makes an enterprise to improve its mobility policy by handling to the organization better control and management of its mobile hardware with simultaneous balancing of investments in UC/PBX systems, office phones and mobile handsets. Abrazo works for a mobile operator, enabling well leveraged high value service offering, which allows them to enlarge their enterprise customer base while minimizing rustle. Tango Networks' innovative hybrid approach to fixed-mobile convergence mixes the service provider and enterprise networks, implementing advantages of the best features of each. "We very are pleased to have Telmar Network Technology offer our Abrazo mobile Unified Communications solution," said Alastair Westgarth president CEO for Tango Networks. "The addition of the Tango Abrazo mobile UC solution to Telmar Network Technology's broad family of systems allows them to have the most advanced mobile UC solutions for maximum customer value. In addition, having Telmar Network Technology provides our customers with spare parts management and out-of-warranty repair service will allow us to have a comprehensive product life cycle offer."

VoIP solutions provide more reliable connection

VoIP solutions provide users with a more reliable connection than standard call networks, according to technology website booshnews.com. The technology uses a packet switched internet which offers multiple paths from any sender to any receiver.

Any fault causing one particular path to close will not result in a lost connection as there is always another line available.

Operators refer to this as a 'self-healing' system, which ensures higher call quality and a more reliable connection than traditional calls.

The website urges businesses to switch to the technology as soon as possible or risk becoming less competitive.

It claimed: "Now is the time for businesses of all sizes to re-examine their phone service, in recognition of the fact that you may be relying on what is already obsolete technology. The time to plan for your changeover is now."

US communications provider AT&T has claimed landlines will soon be obsolete, while around 25 per cent of Americans no longer use the service.

Asterisk High Availability Solutions

Ways to increase system availability and balancing: HAAST from GenerationD www.GenerationD.com offers high availability / clustering for your existing Asterisk servers. A software based solution, High Availability ASTerisk (HAAST) offers rapid automatic failover, manual promote/demote for maintenance, a command line interface, a telnet interface, and a web based interface. Supports master-slave failover, load-balanced, hot-standby, warm-standby modes. Installation is straight forward, with no external hardware required. Real-time fail-over takes 10 seconds to complete. HAAST is a commercial solution, in use at call centers and other high-uptime environments.
DNS SRV on the CPE side but not all phones handle this.
SARK-HA from Aelintra Telecom offers High Availability Asterisk out-of-the box. Runs Aelintra's SARK UCS MVP Asterisk implementation on a pair of servers.... Real-time failover takes less than 20 seconds to complete. Setup requires only 4 additional data fields to filled out in the SARK globals panel. Illustrated set-up guide HERE.
Ranch Networks offers High Availability White_Paper_one_one_HA.pdf solution for Asterisk. This is Hardware based solution. (Just for two asterisks boxes).
Flip1405 Manages virtual IP between two Asterisk servers and queries UDP5060 for state changes Downtime less than 30 seconds Only 2 dependencies (nmap and arping) Incredibly easy to setup
SERVERware is a next generation communication technology solution. Delivers a wide range of IP services and applications. It is redundant, high available and fault tolerant. The SERVER edition of SERVERware is the most economical way to start delivering IP service from a single server. This edition includes all the necessary components that allow service providers to offer any of the supported VPS templates with a clear path upgrade to network edition. NETWORK edition of SERVERware contains up to 256 host servers creating a farm of virtual private servers from which IP service delivery platform is served. This allows service providers to offer redundant, flexible and scalable IP services like mail, web, hosted PBXes etc... Commercial
sysMONIT is a high availability hosts failure detection module that implements small and simple daemon running on each host, sending signaling UDP packets, for purpose of efficient detection of hosts failures, and taking over services by another spare hosts.
Failover switches to automatically switch connections (T1, Ethernet, etc.) to a backup system. Vovida has a SIP load balancer. This allows several Asterisk servers to be setup and appear to be a single server to users. Other load balacing approaches involve the SER SIP proxy, UltraMonkey (see below) or simple DNS round-robin. And then there's also app_distributor as third party application or app_random. there are a lot of bugs and the last version was released in 2002

Use the Linux-HA software to provide high-availability (HA) failover on programmed conditions - by default node hang or crash. Linux-HA also has many telephony-oriented HA APIs as defined by the Service Availability Forum (SAF). It also provides sub-second failover, and works well with shared disk or without. It is commonly used with the DRBD package to provide HA with no single point of failure, and no special hardware requirements.
Stratus, which as been making high-end continuous processing systems for 20 years, has just added an under $10,000 Linux based continuous processing solution: Stratus ftServer T Series Systems


QueueMetrics is able to monitor clustered call-centers with the load distribuited over a number of Asterisk servers as if they were one big single box.
OrderlyStats - Dedicated Real Time Call Centre Management and Statistics Package, can monitor single or clustered asterisk servers from a single page.
Overview
The following is a brief HOWTO for installing High-Availability Asterisk using Open Source tools combined with fail-over capable & intelligent hardware (the fonebridge).
The heartbeat utility is used in a 'Passive-Active' scenario but could easily be modified to do 'Active-Active'.

Background
Some of our more demanding customers in the Call Center and Banking Industry are loathe to accept an implementation with no mechanism for fail-over and high-availability so this is the hardware/software combination we are using to meet their demands.

Client Background
The following scenario was used for a medium sized call center operation with about 60 analog stations, and a single T1 PRI.

Hardware
2 x 1U Supermicro Servers (P4, 512Mb, Dual Gig Eth, Dual SATA with RAID 0) 1 x Redfone Quad T1 fonebridge to terminate PRI connectivity, power channel banks and provide fail-over capability between the two Supermicros. 1 x T1 PRI 3 x Adtran 750 FXS channel banks to drive analog phones 2 x UPS/Surge Protectors
Software
Fedora Core 4 Asterisk, zaptel, libpri from CVS head Linux HA software suite from Ultramonkey. They have RPMs for RHE3 that install fine on Fedora Core 4 Each server is a mirror image of the other in terms of Asterisk configs and software.
Software Install
After a standard install of FC4, Asterisk, zaptel, libpri we installed all of the packages from Ultramonkey pretty much following their guidelines: http://www.ultramonkey.org/3/installation-rh.el.3.html
You may have a few dependencies issues, mainly perl libs, but we were able to satisfy all of them by using Yum. If you are running Apt you should be able to accomplish the same thing.

Configuring Hearbeat
After installing heartbeat there are only three files that need to be modified for your environment. They are ha.cf, haresources and authkeys. They should all be placed in the /etc/ha.d/ directory. The files should be absolutely identical on all machines that are part of your Asterisk high-availability cluster. We only have two servers running but you could easily scale to more using the exact same configurations. These are our config files. All comment lines have been removed but as you can see they are short and simple.

ha.cf
debugfile /var/log/ha-debug
logfile /var/log/ha-log
logfacility local0
keepalive 200ms
deadtime 2
warntime 1
initdead 120
udpport 694
bcast eth0
node asterisk1
node asterisk2

haresources
asterisk1 10.10.10.110 fonulator asterisk

authkeys
auth 1
1 sha1 SuPerS&cretP@$$werd

Operation
Each Asterisk server has a unique IP address which is part of the LAN segment. This could be a NATed network or Internet facing with public IP addresses. Heartbeat manages the monitoring of the hardware state of each machine over Ethernet or serial port or a combination of both (recommended) and assigns the Virtual IP to the Asterisk server which is currently in an active state. Example;

Asterisk1= 10.10.10.100
Asterisk2= 10.10.10.120
Virtual IP= 10.10.10.110 (see haresources)

With Heartbeat it is important that your node names are identical to the host names reflected in #uname -n. You also may need to manually add IP/hosts statements to your /etc/hosts file so each machine knows how to reach the other via IP.

Following the rules in haresources, Heartbeat will assign machine name asterisk1 as the primary server when both systems start up. It will then start the following scripts; fonulator (this is the little script that configures the fonebridge) and asterisk which starts the Asterisk server. These are both standard startup scripts placed in /etc/init.d/ .
If the Primary server suffers a hardware fault or simply stops responding to the heartbeats going between the two nodes asterisk2 will execute /etc/init.d/fonulator start to reconfigure the fonebridge on the fly and begin redirecting traffic to asterisk2 followed by /etc/init.d/asterisk start to start the Asterisk server.

Results
With heartbeat, IP takeover occurs in under a second. The fonulator utility re-configures the fonebridge in just about the same amount of time and then depending on your hardware platform and the complexity of apps running in Asterisk it can take between 5-15 seconds for Asterisk to start up on your secondary server, load all config files, clear alarms and be ready to process calls. Total fail-over time about 15-20 seconds.

Resources
Ultramonkey http://www.ultramonkey.org (High Avail software packages)
Linux HA http://www.linux-ha.org (The High Availability Linux Project)
Redfone http://www.red-fone.com (Maker of the Quad T1/E1 fonebridge)

This tutorial, presented by Open Innovation, aims at presenting a robust cluster architecture to assess reliability based on Open Source software (Postgres, Pgpool, Csync2, ....) and native IP phone features by avoiding complex and expensive common cluster approaches. This solution currently is up and running to serve all business VoIP traffic (800 IP phones) of one of the most used credit card in Italy.
The tutorial is in PDF format and can be downloaded here: A reliable architecture for Asterisk Cluster
Overview
Use standard Ubuntu/Debian packages to create an Active/Passive high-availability solution for asterisk 1.4 using hearbeat 1.0 (and FreePBX) and using SIP (not redphone/PRI/analog/etc). Note: Use Debian server, do not use Ubuntu server until RAID-1 issues are solved (perhaps Ubuntu Intrepid?).

Background
Many ISP's are now providing "Dynamic T1" instead of (or in addition to) standard T1-PRI service. This "Dynamic T1" just means that they are providing highly prioritized VOIP/SIP between your customer site and them across a T1 (or other highspeed connection). So, it is now more and more possible to get cheaper service using VOIP only without T1-PRI and get very similar call quality. This solution deals with Debian/Ubuntu, but also the special issues that are raised with heartbeat when connecting to the upstream provider via SIP. Many clients want failover support to "seal the deal".

Issues
Heartbeat "takes over" an IP address by adding an "alias" to an interface IN ADDITION to an IP that must always be there so that heartbeat can communicate. For a PBX type install that is not behind a NAT, with no upstream SIP proxy (OpenSer), an alias will be added to BOTH the WAN interface and the LAN interface. Asterisk will need to bind to both the LAN and WAN to operate. Unless you do some routing/proxy magic outlined in this solution, you will run into trouble because asterisk will put the wrong SRC/VIA address in IP/SIP packets. This will cause problems upstream, because your ISP/SIP provider may authenticate based on IP and you will be appearing to send packets from the wrong IP. This will cause problems in the LAN for similar reasons.

Software Install
apt-get install asterisk
apt-get-install heartbeat

Heartbeat Config Generally
See the configuration info in the "Redfone" HOWTO above this one generally. I'm using the 10.10.10.0 addresses from above and 77.77.77.0 as a WAN address in my examples. I'm assuming that the shared LAN address is 10.10.10.110 and the shared WAN address is 77.77.77.110. Asterisk1 server's "other" WAN IP is 77.77.77.100. For sake of example: Asterisk2 machine has 77.77.77.120.

haresources
asterisk1 10.10.10.110 77.77.77.110 fixrouting asterisk

Routing fixes
For each interface to which Asterisk binds it gets the IP address by doing a routing lookup. If you look at 'ip route show' and the look after the word 'src' you will see which IP will be used for that interface (also look at 'ip route get'). It will put this IP into VIA headers and send all IP/UDP/SIP packets from this IP. When this server is primary we need to fix the routing so that all packets on LAN look like they are coming from the 'shared' IP of the two servers for the LAN... AND.. (for multi-homed) we need to fix the routing for the WAN interface also.

The 'fixrouting' script detailed below needs to be /etc/init.d/fixrouting
#! /bin/sh -e
set -e

case "$1" in
   start)
ip route change 10.10.10.0/24 src 10.10.10.110 dev eth0
       ip route change 77.77.77.0/24 src 77.77.77.110 dev eth1
   ;;
 stop)
       ip route change 10.10.10.0/24 src 10.10.10.100 dev eth0
       ip route change 77.77.77.0/24 src 77.77.77.100 dev eth1
   ;;
 force-reload|restart)
   $0 stop
   $0 start
   ;;
 *)
   echo "Usage: /etc/init.d/fixrouting {start|stop|restart|force-reload}"
   exit 1
   ;;
esac

exit 0

Results
When a failover happens that makes this server primary the "shared" IPs will be taken over and then the routing fix will make sure that all packets look like they are coming from that IP in asterisk. When this server fails or becomes secondary IPs will be released and the routing fix will set things back to the Passive state so that the Active machine might still be able to communicate with it (and avoid IP conflicts). The current solution I have uses UltraMonkey ( http://www.ultramonkey.org ) for load-balancing and failover and it works like a champ. There are obviously a lot of details there, and I'd be happy to detail them if people are interested. There is also a site that has two clusters with uniform reachability for all phones and PRIs. None of this requires a lot of dialplan tuning on a day-to-day basis.
Asterisk