This talk will discuss how to build a call center with many Asterisk systems across a WAN using an MPLS network. Examples and discussion will focus on a real-life implementation that was built over a period of a year and a half. You will learn about what is possible currently, and some of the limitations in building a distributed call center application. The call center we'll discuss is an inbound call center that utilizes agent logins from SIP devices that can login to queues either in the corporate center, or answer calls from queues located at remote branches. This call center also contains phone level permissions to control who can monitor (or be monitored) and who will have their calls recorded. This is a talk that anyone who is interested in using Asterisk's queue system will want to attend.

Presented by: Leif Madsen VoIP Engineer and Consultant Digium

Presence Technology will show how an Asterisk based all-in-one solution it is the most beneficial alternative for today’s Contact Center needs. Asterisk provides an open communication platform which tightly merged with an all-in-one modular Contact Center solution provides all features needed on a Contact Center with minimum IT burden simplifying the implementation, configuration, and platform operation, all nicely packaged on a unified management interface. There is no need anymore to integrate different and isolated Multi-point contact center solutions which makes tedious to adapt the different systems from a diversity of vendors which needs to be aligned to fulfill the dynamic organizations needs. Using Asterisk drastically reduces the investment needed to own the solution while the all-in-one natively integrated multi-channel suite provides a pre-integrated and fully consolidated Contact Center infrastructure which extremely reduces implementation, maintenance and ownership costs. And what might be most important, all information needed by the operational personnel and decision makers is fully consolidated on a single data repository. At last a solution focused to empower the business user which allows organizations to get focused on their core business.

Presented by: Jose Luis Castaños Founder & EVP of Business Development Presence Technology

SIP trunking has become the next generation of choice for PBX interconnection, and Asterisk fully supports the strategy. For many, the IAX protocol has special merit in PBX trunking and support of services and is overcoming issues with firewalls and network address translation problems. This panel takes a looks at the meet point for interconnection and the considerations needed in developing a strategy to enable carrier-to-business communication.

Presented by: Dayton Turner CTO Voxter Communications

Tired of getting screwed by by your SIP Trunking Providers?

If you resell SIP trunking services from major carriers, then you have probably been shafted multiple times in billing disputes where you paid a lot more than you expected for termination. These settlement disputes often arise because the called number had been ported to a different provider. And the problem is getting worse as the number of ported number grows. The solution to your problem is to route calls based on the routing number, not the telephone number.

This session will provide all the details on arcane telecom details such as the difference between a telephone number and a routing number. This session will also explain how carriers route based on OCN (Operating Carrier Number) and how the LRN (Local Routing Number) relates to the OCN. In addition, we summarize the LERG (Local Exhange Routing Guide) and explain why routing based on NPA-NXX is wrong and why you need to route based on LRN which looks just like an NPA-NXX but is different. In the process, the role of the NPAC (Number Portability Administration Center) will be explained.

To wrap-up we will explain what the Local Number Portability options are for Asterisk users and how they can be implemented.

Presented by: Jim Dalton Chief Executive Officer TransNexus, Inc.

TASterix now supports an integrated Asterisk-based solution (Premise or Hosted) yielding increased flexibility, un-obstructed scalability, and un-matched business continuity while simultaneously offering a unique, customizable, and unprecedented cost structure for the answering service industry.

The term “cheaper” will no longer translate into “Inferior”

History: The answering service industry from an entity standpoint is a very small niche industry with +/- 2000 answering services within the US and Canada. As with any smaller niche industry, software solutions are typically proprietary, more expensive and lag technically behind the typical more aggresive and competitive software industry.

The answering service industry has been almost monopolized by 5-6 software vendors over the past 20 years who consistantly provide legacy phone systems absent of cost reductions passed onto end users.

Conclusion: With the infiltration of Asterisk based telephony, answering service owners will, for the first time in many years, experience a breath of fresh air with more competitive cost structures and more plentiful solutions at their finger tips.

Those answering service software providers who choose not to accept this new Asterisk model both from a cost and architecture standpoint, will only be postponing the inevitable.

The REAL question is, will the other vendors pass on such cost savings to the end users, like TASterix, or will they continue to maintain the old school cost facade which will continue to increase the cost of doing business for the end users/owners.

Presented by: Vince McGlone CEO, President TASterix

One of the most powerful ways to deliver ROI and deliver actionable business intelligence to end-users is by leveraging Asterisk's queue and call detail logging. Whether using a turn-key package like Queuemetrics to provide basic statistics and visibility, or developing full-blown cross-platform custom reports using Jasper Reports--this can take your next Asterisk deployment to the next level.
Session will include:
- Common analytics requests and solutions.
- Discussion of turn-key reporting packages.
- Case studies and reporting examples.
- Ground up analysis of a Jasper Report Server solution.

Presented by: Corey McFadden Managing Partner Infradapt

Software-based solutions that connect to existing telephony resources using Asterisk(r) and the Session Initiation Protocol (SIP) form a complete toolset for efficiently managing your call center and for providing the best possible service to your clients. Intelligent software applications that provide directory information, on-call scheduling, integrated messaging and automated contact-based dispatching maximizes return on investment by streamlining and auditing the call flow from start to finish. Client desktop communication tools make it possible to extend the power and functionality of these intelligent software applications directly to customers with no operator involvement. Remote operators have complete access to the same tools and capabilities as those in the call center through the software-based nature of these SIP solutions. AMTELCO is a Digium Asterisk Interoperability Partner, has been an industry leader in providing innovative call center solutions for more than 30 years, and has been awarded more than 20 U.S. and foreign patents. Attend this session and see the future of call processing, data capture, multi-database integration, reporting and more through AMTELCO's Intelligent Series software solutions.

Presented by: James Kleckner Product Manager AMTELCO

Every VoIP provider can become a Mobile VoIP provider.

You'll learn how to reduce customer provisioning costs and sell more minutes.

We'll discuss:

* the advantages of using the customer's mobile phone as CPE;
* how mobile VoIP translates into more VoIP minutes sold;
* the ways of offering mobile VoIP to your customers.

Presented by: Eric Chamberlain CEO/Founder RF.com - RingFree Mobility Inc.

This talk will cover how to ensure that your production carrier telephony services run reliably. Topics include:

* Management culture.
* Cluster architectures.
* Rolling out upgrades and configuration changes.
* Staff responsibilities.
* Dealing with vendors.
* How to grow system capacity.
* Causes of outages in the real world.

Advice will be based on real-world experience from supporting 45 Enswitch systems of up to 150,000 users. There will be very little theory and lots of specific advice. If time permits, there will be a question and answer session at the end.

Presented by: Alistair Cunningham Managing Director Integrics Ltd

Staring with version 1.5.0, OpenSIPS has the ability to perform real load balancing between heterogeneous Asterisk peers. Each Asterisk can provide different sets of resources (like voicemail, transcoding, gatewaying, conferencing, etc).

For a call requiring a set of resources, OpenSIPS can determine which is the Asterisk box that can compete the call, keep the level load on the system.

The load-balancing mechanism is dynamic, can be tuned during runtime and also peers can provide feedback to the load-balancer in regards to their current load or changes in capacity.

http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing

Presented by: Bogdan-Andrei Iancu Founder and developer OpenSIPS project

This session will go over examples of how to make text-to-speech applications using dynamic and pre-recorded audio prompts, including multi-step IVR text-to-speech applications.

Presented by: Matt Florell President Vicidial