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Call forwarding/redirection in FreeSWITCH

9 Mar

Consider you have two different contexts in your dialplan for inbound and outbound calls: the “public” context transfers the calls into “XXX_inbound” (XXX being your organization name), and the user directory has “XXX_outbound” as “user_context” variable.

Having two contexts, you have more flexibility in defining the short dial strings and outbound destinations.

But there’s a little problem: if the SIP client redirects the ringing call, or if the user makes an attended transfer, FreeSWITCH would initiate a new outbound leg in the same context where the call was bridged toward the SIP client.

As a solution, you need to define a new extension in your XXX_inbound context which would match PSTN outbound numbers. The channel will already have all custom variables which were set before bridging toward the SIP client, so you can set an additional condition criteria to make sure that this is the redirected call. This example would be placed at the bottom of the inbound context, and “directory_ext” is the variable that was earlier in the same context before the call was bridged to the SIP client:

    <extension name="call_forward">
      <condition field="destination_number" expression="^\d+$"/>
      <condition field="${directory_ext}" expression="^70\d$">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=false"/>
        <action application="bridge" data="${outgw}/${destination_number}"/>
      </condition>        
    </extension>

-
Source: http://txlab.wordpress.com/2014/02/16/call-forwardingredirection-in-freeswitch/


Installing PBX debug tools in RHEL v6 (Asterisk v1.10+, FreePBX v2.10+)

10 Feb
PBX,(Private Branch exchange) is a private telephone network used in mid-size enterprises. Consumers that use PBX are configured in a particular number of outside lines to make phone calls for the PBX. Companies make use of a PBX because it’s much less expensive connecting an outside phone line to every single telephone from the organization. It’s much easier to call someone in a PBX because the total numbers you need to dial is usually 3 or 4 digits.

You only require to run in console text mode not GUI graphics mode.  If you currently have a desktop or server GUI installed you are going to need to exit to console mode. you need to automatically let your server boot into a console. You do that by typing init 3 in your inittab file from the terminal console. You will have to have root privileges. Eliminate all installed groups except ‘Yum Utilities’ and then you can start confirming the delete list before entering ‘y’ to make sure none of the ‘sshd’ or ‘yum’ they check change with newer revisions.

yum grouplist installed

Installed Groups:
  DNS Name Server
  Editors
  Legacy Network Server
  Mail Server
  Network Servers
  System Tools
  Text-based Internet
  Web Server
  Windows File Server
  Yum Utilities
DNS Name Server'
  yum groupremove 'Editors'
  yum groupremove 'Legacy Network Server'
  yum groupremove 'Mail Server'
  yum groupremove 'Network Servers'
  yum groupremove 'System Tools'
  yum groupremove 'Text-based Internet'
  yum groupremove 'Web Server'
  yum groupremove 'Windows File Server'

update the base install

DNS Name Server'
  yum groupremove 'Editors'
  yum groupremove 'Legacy Network Server'
  yum groupremove 'Mail Server'
  yum groupremove 'Network Servers'
  yum groupremove 'System Tools'
  yum groupremove 'Text-based Internet'
  yum groupremove 'Web Server'
  yum groupremove 'Windows File Server'

update the base install

yum -y update

Install Asterisk/FreePBX required packages, other useful packages, with their dependencies

yum groupinstall core
yum groupinstall base
yum install gcc gcc-c++ wget bison mysql-devel mysql-server php php-mysql php-process php-pear php-pear-DB php-mbstring nano tftp-server httpd make ncurses-devel libtermcap-devel sendmail sendmail-cf caching-nameserver sox newt-devel libxml2-devel libtiff-devel php-gd audiofile-devel gtk2-devel subversion nano kernel-devel selinux-policy sqlite-devel.

Since php-pear-DB package isn’t included with RH linux version and its clones, you can download it from an official mirror and install otherwise the FreePBX install will fail.  update it to the most recent version

cd /usr/src

wget http://dl.fedoraproject.org/pub/epel/6/i386/php-pear-DB-1.7.13-3.el6.noarch.rpm

rpm -ivh php-pear-DB*

Firewall

Find out if the firewall (iptables) is activated automatically and if the RHEL v6 default configuration blocks the FreePBX web GUI.  Once you know what services/ports are essential you’ll be able to run “system-config-firewall-tui” and configure the firewall.

To start, these ports need to be opened:
TCP 80 (www)
TCP 4445 (Flash Operator Panel)
UDP 5060-5061 (SIP)
UDP 10,000 – 20,000 (RTP)
UDP 4569 (IAX)

An alternative option is usually to remove existing settings from your firewall and save.

iptables -P input accept
iptables -F
service iptables save

you’ll also be able to disable the firewall for the present time and prevent it from starting on reboot.

service iptables stop
chkconfig iptables off

Selinux

Selinux isn’t needed or recommended at this point. So you need to disable selinux from “enforce” to “disable”

nano /etc/selinux/config

# This file controls the state of SELinux on the system.
# SELINUX= can take one of these three values:
#       enforcing - SELinux security policy is enforced.
#       permissive - SELinux prints warnings instead of enforcing.
#       disabled - SELinux is fully disabled.
SELINUX=disabled
# SELINUXTYPE= type of policy in use. Possible values are:
#       targeted - Only targeted network daemons are protected.
#       strict - Full SELinux protection.

SELINUXTYPE=targeted
# SETLOCALDEFS= Check local definition changes
SETLOCALDEFS=0

(Ctrl-x> y >Enter)

Turn off selinux for any session

setenforce 0

TFTP

(Enable the TFTP for configuring phones)

nano /etc/xinetd.d/tftp

change server_args = from “-s /var/lib/tftpboot” to “-s /tftpboot”
change &ldquodisable=yes&rdquo to &ldquodisable=no&rdquo

(Ctrl-X>y>ENTER)

mkdir /tftpboot
chmod 777 /tftpboot
service xinetd restart

Set Timezone

System timezone
Create the appropriate timezone from /etc/localtime.
Example:

ln -sf /usr/share/zoneinfo/America/Vancouver /etc/localtime

PHP Settings

To avoid a variety of warnings set, PHP to the the correct default time zone

nano +946 /etc/php.ini

Uncomment () date.timezone = add your time zone

Adjust memory limit accordingly

nano +457 /etc/php.ini

memory_limit = 128M

restart apache or your installed websever then afterwards to effect your changes

service httpd restart

Get FreePBX

Check if this is the latest released version.

cd /usr/src wget http://mirror.freepbx.org/freepbx-2.10.0.tar.gz tar zxvf freepbx-2.10.0.tar.gz

Get and Install Asterisk

wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-10-current.tar.gz tar zxvf asterisk-10-current.tar.gz

cd /usr/src/asterisk-10*

make clean && make distclean

./configure && make menuselect

After installation create a user

useradd -c “Asterisk PBX” -d /var/lib/asterisk asterisk

and lastly set ownership to these files

chown -R asterisk /var/run/asterisk
chown -R asterisk /var/log/asterisk 
chown -R asterisk /var/lib/asterisk/moh

then change the apache group and users

sed -i “s/User apache/User asterisk/” /etc/httpd/conf/httpd.conf

sed -i “s/Group apache/Group asterisk/” /etc/httpd/conf/httpd.conf

MySQL Setup

First, discover if mysql instance is running, if not start it manually

service mysqld start

Initializing MySQL database:                               [  OK  ]
Starting MySQL:                                            [  OK  ]

configure the mysql database  using -p option if you have an SQL password as MySql user, FreePBX will prompt you for the password  mysqladmin -p create asterisk

cd /usr/src/freepbx-2.10.
mysqladmin create asterisk
mysqladmin create asteriskcdrdb
mysql asterisk < SQL/newinstall.sql
mysql asteriskcdrdb < SQL/cdr_mysql_table.sql

Securing any databases is mandatory and in a case where your server does not have an active firewall, then you need to set  bind-address = 127.0.0.1  so that MySQL  listens only to localhost and nothing else. If you do not perform this step, anyone can use your settings or credentials via the HTTP (80 port if you are using gui to access the administrators panel) and gain administrative access. For security reasons a database password should be strong and consist other characters than the normal alphanumeric; for this session we will use a simple username “admin” and password “admin” using mysqladmin -u root password ‘admin’ command.

mysql

mysql> DROP DATABASE test Query OK, rows affected (.00 sec)

mysql> SHOW VARIABLES LIKE 'hostname' +---------------+----------------+
| Variable_name | Value          |
+---------------+----------------+
| hostname      | somehostname.com |
+---------------+----------------+
1 row in set (.00 sec)

mysql> DROP USER ''@'localhost' Query OK, rows affected (.00 sec)

mysql> DROP USER ''@'somehostname.com' Query OK, rows affected (.00 sec)

mysql> DROP USER 'root'@'somehostname.com'

mysql> GRANT ALL PRIVILEGES ON asteriskcdrdb.* TO admin@localhost {IDENTIFIED BY|Recognized By} 'admin' Query OK, rows affected (.00 sec)

mysql> GRANT ALL PRIVILEGES ON asterisk.* TO admin@localhost {IDENTIFIED BY|Recognized By} 'admin' Query OK, rows affected (.00 sec)

mysql> flush privileges Query OK, rows affected (.00 sec)

mysql>q Bye

install FreePBX now

cd /usr/src/freepbx-2.10.0

./install_amp

set FreePBX to start on boot

echo /usr/local/sbin/amportal start >> /etc/rc.local

Enable Apache and MySQL to start on boot


chkconfig httpd on
chkconfig mysqld on

After a  reboot  you should be able to access

Steps you need to take after reboot

You can access the graphical login of FreePBX with your web browser in the format http://IPaddressOFyourFreePBXserver/.

When accessing web-based FreePBX Admin GUI for the first time you need to “Apply Configuration Changes” so all the *.conf files are installed. or it can also be achieved  by using the command ‘amportal restart’ from the console.

when adding the external SIP extension in FreePBX, change the nat=never default in the configuration to nat=yes for the extension that will be external.

delete or rename /etc/asterisk/sip_notify.conf when you get a symlink fail error during FreePBX installation

Setup external sip extensions if going through NAT

nano /etc/asterisk/sip_nat.conf
nat=yes
externip= or
;externhost=yourdns.com
localnet=192.168.1.0/255.255.255.0
;change the above to whatever your local subnet is
externrefresh=10

(Ctrl-X>y>ENTER)

 

Source: http://edzeame.wordpress.com/2014/02/07/installing-pbx-debug-tools-in-rhel-v6-asterisk-v1-10-freepbx-v2-10/

Three Essential Telephony Services for Your Business

20 Jan

If you look back a few decades ago, the average corporate phone system was rather simple, involving a switchboard and several desk phones around the workplace. These days though, new technological developments have combined with greater competition to create an environment in which the simple telephony service of the past has evolved into a very different character. If you’re confused about which options to take, here are three of the best to benefit your business.

pbxHosted IP PBX Services

In the past, a large business would have to invest in an expensive private branch exchange (PBX): a system of telephones, switchboards & cables that ran throughout the office. These required external contractors for repair and maintenance as the typical corporate employee wouldn’t have the technical experience.

Needless to say, this was a bulky, inefficient option that has now been replaced by the hosted IP PBX service which requires no setup costs or technical expertise. With these packages, your provider has already established a state-of-the-art system on their premises. For a monthly fee, you can tap into this and gain a fully functional communications platform without any large installation and maintenance costs. These systems also offer other features such as:

  • Unified messaging
  • Softphones

This is the ideal solution for anyone seeking a better corporate telephone system without the hassle and cost of installing one directly onsite.

Portrait of customer service operators communicating in a call center

Call Answering Providers

When running a company, you’ll also have to determine how to successfully handle your incoming calls throughout the day and night. While you could hire a team of fulltime receptionists, the resultant salaries can be quite costly. A better solution is to outsource these services to a locally based agency instead. The good news is that anyone can find telephone answering packages tailored to their company. In this way, you can hire a third party to take your excess calls whenever you need it. They will be fully trained in your preferred manner, systems and corporate details too, allowing you to handle all incoming calls in the following situations:

  • Talking to customers after hours and on weekends
  • Temporarily replacing receptionists who call in sick
  • Hiring additional staff during high volume call periods

Thus, you can take important customer messages regardless of what’s happening without any reduction in your quality of service or client satisfaction.

voip

Voice over IP (VoIP) Packages

The last essential service is the VoIP package, an option that allows you to make calls over the internet instead of through the traditional phone line. Not only will this save you money when it comes to overseas or interstate calls but the call quality will also be a lot better especially if you’re using fibre optics or a 4G data connection. If you’ve chosen an affordable call handling package for your business, you should then have enough money to consider a VoIP phone system and call plan as well! Of course, you’ll have to consider the following options before you sign the contract:

  • The cost per month or per call of the package
  • Whether you can call for free to any devices or apps
  • The change in rates for long distance phone calls
  • The type of softphone that you need to install
  • The compatibility of these with your current setup

Just think carefully and compare what’s on offer, and you should then be able to choose a VoIP provider and package that suits your corporate needs perfectly.

These three services are absolutely essential for the modern business. If you’ve yet to experience their benefits, we recommend you get in touch with the appropriate specialists as soon as possible. The success of your business depends on this!

 

Source: http://nancy-rubin.com/2014/01/16/three-essential-telephony-services-for-your-business/

Traditional Telephony No Longer Cuts It

28 May

Business telephony has evolved in silos segmented by location, network and business need. The result is over complicated, overpriced and inflexible business communications at the heart of most businesses.

Traditional Office Based Telephony
The standard approach to delivering business telephony is the PBX (Private Branch Exchange). In our head office we install one of these, place often proprietary telephones on each employee’s desk and connect to the outside world through ISDN lines. If we have regional or branch offices we replicate this on a smaller scale.

This traditional approach leaves us with a distributed architecture with multiple platforms in multiple locations, with little to no seamless connectivity between sites – many organisations still have to dial between sites using the PSTN.

Additional Requirements, Additional Boxes
The number of silos increase when there is a need for additional capability. For example, I want to support a small contact centre with the traditional approach- this means a separate box with separate handsets and separate connections to the PSTN, another silo.

The same is true if I want to add functionality to my existing telephony. For example, I want some form of auto-attendant, IVR or voice recording. I have to add a box to each PBX or site which is a lot of duplication and unnecessary expense.

The Separate Worlds Of Fixed & Mobile
Two very separate and distinct worlds of communications have emerged because we are using business mobiles more often than ever before.

How many times do you call someone’s office number only to get voicemail? You leave a message and instantly try their mobile – surely the solution is ‘one number reach’ regardless of device.

The Cloud Has To Be The Answer
The approach we have taken in computing- to centralise applications and deliver their capability to anyone, anywhere at any time- has to be the answer for business communications.

By centralising our communications in the cloud, we create a single platform that serves and unites all users.  The cloud allows us to scale and extend this platform to support remote workers and home workers resulting in one system for every employee.

It provides the ability to turn on additional functionality to meet business need such as IVR self-service, auto-attendant, contact centre or recording and to do this in a single place, available to everyone.

In addition it also gives us the ability to converge fixed and mobile communication. If there is no answer from a user’s desk phone, the call is automatically routed to their mobile, providing single number reach. The caller does not have to hang up and redial the mobile number.

This approach transforms business communication, achieving economies of scale through centralisation, simplifies management by having a single platform and provides the ability to deliver the capability each individual needs, when they need it, wherever they need it, and on the device of their choice.

In terms of business telephony, the cloud most definitely has the silver lining.

Source: http://cirrusresponse.wordpress.com/2013/05/28/traditional-telephony-no-longer-cuts-it/

Advantages to Customers of SIP Trunking

23 Apr

SIP stands for Session Initiation Protocol and is a technology at the enterprise level for delivering multiple voice connections to a PBX or key system over an IP data connection. In order for a business to utilize SIP they must have a PBX with a SIP-enabled trunk side and their data provider must be able to deploy and switch SIP.

Hot Desk, GTi, University of GlamorganSIP Trunks at the enterprise level of the network replace PRIs between the central office andPBXs. A PRI is a dedicated T-1 transport circuit and can support 23 bearer paths for voice, but aSIP trunk connection typically rides an existing data circuit and can be used to carve out as many voice paths as are wanted within the limits of the bandwidth available.

Following are the reasons that businesses want SIP trunks, and thus for carriers to sell them. This list is discusses the advantages for the small and medium business customer.

Saves Money. SIP generally saves money. SIP trunks replace PRIs which are inefficient. It is not unusual for a customer with a PRI to be using only part of the capacity and yet they have to pay for it all since it is a linear product. SIP trunks are typically carved out of a company’s data or Internet connection and can be sized as needed within the constraints of the bandwidth. It is typical for a business to cut their costs at least in half using SIP trunks compared to PRIs due to the efficiency.

More Efficient Use of the Data Connection. Most businesses will already have an Internet connection and SIP trunks are carved from those connections. Most businesses use their data connections in a bursty fashion, meaning there are times of the day when they use a lot of their bandwidth, but also many times when they use very little. SIP trunking can take advantage of the unused capacity in most company data connections. Companies often do not need to increase the bandwidth they are buy SIP trunks and can fit them into their existing data product.

Enables Unified Communication. SIP enables all of the various features that comprise unified communications such as access to the phone system from cell phones or tablets, integrated voicemail and email, video chat, instant messaging and other features that make businesses more productive.

Enables Upgrade to an IP PBX. Businesses more and more want the kinds of features that are available with an IP PBX and IP handsets. Many businesses are choosing to buy an IP PBX to get these features rather than buy IP Centrex from their telco provider. The general advantage for a business to have their own IP PBX is the ability to customize their communications network, something that many service providers do not offer with IP Centrex.

Allows Multiple Locations to Act like One. With SIP trunks and an IP PBX a business with more than one location can have a unified telephone system that brings the data and voice together for all locations.

Any carrier that sells enterprise data service to businesses should offer SIP trunks. Even if you sell IP Centrex, customers who prefer to have their own phone system are going to want SIP trunks.

Source: http://potsandpansbyccg.com/2013/04/23/advantages-to-customers-of-sip-trunking/

Traditional Phone PBX vs VoIP Phone [Infographic]

14 Mar

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The Technology Report March 2013

8 Mar

This report provides a glimpse of the market dynamics, trends, what’s new and some insight into areas of opportunity in the IP PBX / Enterprise Telephony sector. The following is from analyst firms including Dell’Oro Group, Infonetics Research, IDC, Ovum, and CollabTel / CXO Reports.

The following is from the Dell’Oro Group based on Q2 & Q3 2012 from approximately 40 OEM’s:

IP   PBX Large

Sequential   Growth

Quarter   2 2012 / Quarter 3 2012

Revenue ($M)

$283.8

1%

Lines (000’s)

2,540.7

8%

Units (000’s)

14.3

9%

IP   PBX Small

Revenue ($M)

80.2

10%

Lines (000’s)

919.8

10%

Units (000’s)

40.7

9%

During 3Q12, total PBX revenues increased 3% sequentially to $1.2B, along with Enterprise Telephony which also increased 3% sequentially to $3.0B. Enterprise voice application revenues grew 5% sequentially to $745M.

As always, I want to thank the folks at The Dell’Oro Group for their terrific support in furnishing the primary source of information above (and below).

________________

IP Multimedia Subsystem Core Revenues Grew Over 80% Year-Over-Year, According to Dell’Oro Group

Beneficiaries Include Alcatel-Lucent, Ericsson, Huawei, Mavenir and Nokia Siemens

REDWOOD CITY, Calif. – March 7, 2013 – Dell’Oro Group, the trusted source for market information about the networking and telecommunications industries, announced today that while the overall Carrier IP Telephony market revenues declined 5% versus the year-ago period, sales of IP Multimedia Subsystem (IMS) Core systems were through-the-roof.  IMS sales were driven by operator capital spending on current as well as planned launches of the new Voice Over Long Term Evolution (VoLTE) service.

“VoLTE is creating both opportunities and disruption in equipment spending patterns. We expect a significant increase in VoLTE services in 2013, with as many as twenty operators carrying native Internet Protocol voice from handsets by year-end,” said analyst Chris DePuy.  “While IMS and Wireless Application Server spending accelerated in 2012, we expect further spending will be stimulated by the availability of VoLTE-capable handsets in the next several months.  Spending on certain wireline voice projects has been postponed as operators prioritize on wireless voice systems upgrades.”

The overall Carrier IP Telephony market, which includes devices used to serve both circuit switched subscribers and Voice Over IP (VoIP) subscribers, reached revenues of just under $2 billion.

________________

Infonetics: PBX Revenue Falls Despite Demand; Unified Communications Vendor Battle Looming

Market research firm Infonetics Research released vendor market share and preliminary analysis from its 4th quarter 2012 (4Q12) and year-end Enterprise Unified Communications and Voice Equipment report: (Full report published March 4).

“Following two years of modest growth, the PBX market had a tough 2012. While shipments grew, competitive pricing pressure persisted, driving down worldwide revenues, exacerbated by Europe’s tough economic conditions,” notes Diane Myers, principal analyst for VoIP, UC, and IMS at Infonetics Research.

“On a brighter note, unified communications ended the year on a high note, led by Microsoft and its Lync platform,” Myers continues. “We expect the PBX market to move back into positive territory in 2013, with moderate growth in addition to continued strong adoption of UC applications by large and mid-market enterprises looking to improve flexibility and productivity.”

ENTERPRISE TELEPHONY MARKET HIGHLIGHTS

  • The global PBX/KTS market totaled $8.1 billion in 2012, down 4% from 2011; the EMEA region alone declined 10%
  • In 2012, the unified communications segment grew 8% year-over-year
  • North America and APAC were the regions to net positive revenue gains in the PBX market in 2012, growing a modest 2% and      1%, respectively over the previous year
  • Pure IP PBX licenses grew 11% in 2012
  • Cisco maintains its lead in global PBX/KTS revenue for the 6th consecutive quarter; Avaya and Siemens round out the top 3
  • Only 3 vendors in the enterprise telephony market posted year-over-year revenue gains in 2012: Cisco, Mitel, and ShoreTel
  • Infonetics expects fierce vendor battles to continue in 2013, particularly between Microsoft and Cisco in the unified communications segment

ViewMedia_359387_original

Cisco, Avaya, Siemens and NEC are the top PBX and unified communications equipment vendors for the full year 2012, reports Infonetics Research

______________

EMEA Unified Communications and Collaboration Market Expected to Reach $11.7 Billion by 2016, Says IDC

MADRID, December 17, 2012 — According to a new study from International Data Corporation (IDC), the unified communications and collaboration (UC&C) market in EMEA will be worth around $6.9 billion in 2011 and $11.7 billion by 2016.

“IDC predicts a compound annual growth rate of around 10.9% in the next five years and believes that though we have seen moderate growth in the EMEA region, the recession will continue to have a knock-on effect on new shipments of UC&C technologies throughout the forecast period,” said Isabel Montero, senior research analyst, IDC EMEA Unified Communications and Collaboration. “We don’t anticipate an immediate recovery, especially if the eurozone were to dissolve. The current state of the economy is diverting businesses from making investment decisions in new IT technologies and as a result the cost versus ROI of such investments will be a top priority.”

The study also reveals that:

  • Of the major economies in the region, only Germany achieved double-digit growth in UC&C deployments      in 2012 compared with 2011. Germany grew 10.3%, and was followed by France (9.6%) and the U.K. (7.7%).
  • Russia, the Czech Republic, and Poland are expected to have the strongest CAGRs in the Central and Eastern Europe (CEE) region throughout the forecast period. This will be primarily driven by the replacement of traditional equipment with next-generation enterprise voice connections.
  • South Africa and Turkey will continue to be the leading countries in the Middle East and Africa (MEA) region in terms of higher deployment levels of UC&C technologies. Companies in these countries are starting to invest heavily in the replacement of existing infrastructure in order to be able to integrate UC&C technologies, with a special focus on videoconferencing equipment and collaborative applications.
  • Although we have seen moderate growth in the EMEA region, the ongoing recession will have a knock-on effect on new shipments of UC&C technologies throughout the forecast      period. The need for businesses to reduce operational costs (such as travel) will drive demand for collaborative applications such as conferencing applications, enterprise social software platforms, and      videoconferencing / Telepresence equipment.

One of the key challenges for organizations looking to adopt UC&C is identifying the right mix of UC&C technologies, features, and applications that are the most appropriate for their business needs and, more importantly, are the sources they would consider for UC&C deployments and expertise. “An equally important consideration, especially among larger organizations, is whether or not to develop a UC center of excellence [COE] within the organization,” added Montero. “This would be a designated group that brings together individuals from various areas of the organization — such as IT, business applications, lines of business, and customer services — to provide guidance and direction for UC&C deployment plans, project development, and ongoing usage. This group could be formed early on, ideally to gain experience and feedback from initial projects.”

______________

Snippet from a recent 24-question vendor survey on Cloud Migration[1]

Question: What is the impact on channels for marketing, integration and customer support?

Interactive Intelligence’s Response:

Consistent with our premise business, our go-to-market strategy is to sell through a strategic mix of direct and indirect channels. We have two options for indirect partners to sell our cloud offering:

1. Referral partner – the partner identifies an opportunity, does the initial presentation, and turns the opportunity over to Interactive Intelligence for a referral fee.

2. Integration partner – as above, the partner handles the initial steps in the sales process. They then engage Interactive Intelligence in a joint selling effort and work collaboratively with us to ensure a successful implementation and customer experience.

______________

Trends

IDC Worldwide and U.S. Enterprise Mobility Network Consulting and Integration Services 2012–2016 Forecast

Abstract

“Services firms provide the essential guidance that allows enterprises to extract maximum business value out of their infrastructure investments while positioning them for future growth, agility, and competitiveness. Demand for enterprise mobility network consulting and integration services will grow at a pace of 18.4% over the next five years, driven by the enterprise’s desire for increased productivity and collaboration along with the ability to access data anytime anywhere in a secure fashion,” said Leslie Rosenberg, research manager for IDC’s Network Life-Cycle Services program.

_______________

What’s New:

Ovum finds Networking Start-ups Overdue for VC Rebound

London, 26 September 2012 – Since the financial crisis, venture capital (VC) firms have been giving much of their tech cash to mobile, social, and OTT start-ups, showing little interest in telecom. VC support for telecom infrastructure start-ups has dropped from US$796m in 2009 to just US$270m in the 3Q11–2Q12 period. In a new report, though, global analyst firm Ovum finds reason for optimism, concluding that “recent IPO and M&A transactions point to a rebound in VC interest in network infrastructure.”

While VC support for network infrastructure has declined, overall VC investments have recovered, growing from US$20.1bn in 2009 to US$27.8bn in the four quarters ended 2Q12. Some of the beneficiaries of this modest surge include Facebook, Groupon, Twitter, LivingSocial, Square, Lashou, Kabam, WhatsApp, and Spotify.

Matt Walker, Ovum Principal Analyst and author of the report, explains:  “A funding disconnect has thereby emerged between network builders and network users. Lots of innovation and venture capital is targeting the network users, such as mobile apps and OTT platforms. However, little of it is directly helping the network builders. With a weak start-up pipeline, the industry relies more on incumbent vendors to generate new ideas and products. Their budgets are bigger, but VCs are often better at funding ‘game changing’ ideas ignored by established vendors.

“Incumbent vendors’ internal R&D budgets are now nearly 90 times larger than VC investments in the sector, up from 30 times two years ago. This narrows options for service providers, who rely on both large and small vendors for innovation. The big vendors also need access to the start-up pipeline, to fill in gaps in their own portfolios through partnership and M&A.”

In response, service providers are getting more actively involved in funding and working with start-ups. Telefonica, Vodafone, Verizon, AT&T, KDDI, China Mobile and many others are now funding start-ups directly, often deploying products in the network or lab ahead of commercial availability. Earlier this month, Deutsche Telekom (DT) was the latest carrier to announce a new push on the venture side, revamping its T-Venture unit to foster purchase of majority stakes and accelerated disbursement of funds.

Walker adds: “Carriers really need help from suppliers, yet what they face is a vendor market in confusion. Most large vendors are now shrinking and reorganizing, even the Chinese suppliers. Several vendors are modifying business plans and selling assets in order to stay solvent. With the recent VC drought in networking, it’s not surprising that big telcos have become more directly involved in funding start-ups.”

Walker points out that, based on data from the PWC/NVCA MoneyTree Report, the “Networking & Equipment” share of total VC investments shrank to just 1.0 percent for the past four quarters (3Q11–2Q12), down from about 10 percent in 2003.

The good news is recent IPO and M&A deals do suggest that VCs are looking favourably on the telecom sector again, when telecom is defined broadly. For instance, VC-funded start-up Nicira Networks was recently acquired by VMware for US$1.26bn. “The tide seems to be shifting. With heightened investor interest and carrier need for solutions in such areas as small cells, network virtualization, and network optimization, telecom network infrastructure VC seems ripe for a rebound,” concludes Walker.

 

Source: http://www.telecomreseller.com/2013/03/08/the-technology-report-march-2013/

PBX replacement with MS Lync (with Dual Forking) Part 2

9 Jan
As i mention in my last post (part 1) we can choose to use the Voice gateway in pass-through as shown here :

PBXReplacementPart2_Example_base scenario

but to do this , we have to consider various steps before moving the IP-PBX in production  and insert voice gateway between PSTN and the enviroment, so let start from the beginning,  these are necessary steps :

1.  Voice gateway configuration for PSTN trunk (only configuration not yet connected), in the other side (to lync and to PBX) configuration of SIP trunk to Lync and SIP trunk to IP-PBX (IMPORTANT : to obtain a dual forking of calls in this scenario we must use only sip trunk to and from  PBX , so we must have an IP-PBX with sip trunk enabled ).

At this step we don’t have any disservice on IP-PBX production environment but we are ready to switch PSTN from IP-PBX to Voice gateway .PBXReplacement_part2_Example_step1

2.  We can schedule session tests to verify that all configuration made before work    fine (for example during time range in which we couldn’t have any outages to users, for example during the night?), in this way if not all scheduled tests list will be fine , we can rollback easily and move PSTN trunk to IP-PBX again.

To achieve this we must locate the Voice gateway properly sized, to mantain compliance on  requirement in terms of business continuity, for example redundant power supply , right number of SIP channels to/from lync, and to/from IP-PBX.

In this way we can also test a SIP trunking from a provider instead of PSTN (or buy another PSTN trunk to do a test pilot for Lync voice for example) , because until we switch the PSTN from IP-PBX to Voice Gateway, we can work easily in Voice Gateways side without give any problem in IP-PBX side.

About call flow management and dual forking we have the same behavior as i wrote in my previous post (part 1) , unique difference is that all configuration for dual forking is made in Voice gateways side , and in PBX side we have only to switch all inbound and outbound call to the new SIP trunk instead of PSTN trunk already switched off.

At this point we can assert that we have a lot of way to do a soft migration or simply to use the existence telephony infrastructure for Lync and generally Microsoft UC. We know that Microsoft Lync just for presence/audio/video/conference is really wasted and with a good approach and a small effort we can implement an excellent Lync Voice project for a really Unified Communications experience.

Source: http://msucblog.wordpress.com/2013/01/08/pbx-replacement-with-ms-lync-with-dual-forking-part-2/

PBX replacement with MS Lync (with Dual Forking) Part 1

9 Jan
Talking about PBX replacement with MS Lync can be a difficult argument when proposed to customers. But as the nature of MS Lync we have a lot of ways to do it. Usually we can meet two different type of customers, one can think that his employees must change how they work day by day, and for this reasons we can explore solution with direct switch to new technology providing a direct cut-off ; the other one,  not so confident,  prefer a soft migration and possibly a true coexistence between old and new phone system, the last one obviosly is more complicated,  but surely the most funny for us:-), i want to explain  you how we can do a soft migration also with a good coexistence, for now i can mention 2 type of IP-PBX or TDM-PBX: ALCATEL OXE and Cisco CCM.

The first important thing is that all of this project must provide a Voice Media Gateway to ensure that all translation and, eventually transcoding,  from one system to Lync and viceversa don’t drive us crazy…:-)

Actual Infrastructure Enviroment without integration

PBXReplacementExample_base scenario

Based infrastructure consider that we have a fully up and running Lync enviroment and a consolidate Phone infrastracture .

Scenario  (Coexistence with Dual Forking)

If we have a ALCATEL OXE (with remote extension license), CISCO CCM (sip forking with extension mobility license) or a TDM/IP-PBX that support forking to another number not included in its dial-plan (for example to a sip trunk or TDM trunk connected) we can consider this scenario :

PBXReplacementExample_scenario1

Using  Voice Gateway between Lync and Phone infrastructure give us a lot of configuration that otherwise we could not easily do without provide a big effort from the Phone system team .

In this scenario we can consider this events :

Inbound call from PSTN : When we receive a call from PSTN to +3906….4444 , call arrive to PBX , PBX at this stage send the call to the extension in its dialplan and see that there’s also another number associated (for example 9994444) and , in parallel , fork this call to that number with 999 (a prefix trunk associated to the Voice gateway).

When the call arrive to Voice Gateway with destination number 9994444 , it translate called number in +3906….4444 and send to lync .

Result  :  Lync client (or lync phone) and PBX phone ring at the same time, and when one of this two pick up the call ,the other one stop ringing .

Inbound call from another PBX phone : When we receive a call from another PBX phone  to 4444 , call arrive to PBX , PBX at this stage send the call to the extension 4444 and see that there’s also another number associated (for example 9994444) and , in parallel , fork this call to that number with 999 (a prefix trunk associated to the Voice gateway).

When the call arrive to Voice Gateway with destination number 9994444 , it translate called number in +3906….4444 and send to lync .

Result  :  Lync client (or lync phone) and PBX phone ring at the same time, and when one of this two pick up the call ,the other one stop ringing .

Outbound call from Lync to other PBX phone : In lync we have two way to make a call to a contact, if we make a classic Lync call , this call remain inside Lync enviroment but if we make a work phone the call is translated for example in extension format :

– Digited : +3906……4444 , normalized in 4444  so the call are sent outside Lync through the Voice Gateway  and arrive to the extension 4444 , as i mention before in pbx enviroment 4444 have another extension configured (9994444) that corrisponds to the Voice gateway trunk and the same call was also diverted to Lync client .

Result  :  Lync client (or lync phone) and PBX phone ring at the same time, and when one of this two pick up the call ,the other one stop ringing .

Yes i know , a little cumbersome but it’s work fine .

Inbound call from lync to PBX and dual forked to lync

Outbound call from Lync to other PSTN: All external calls made from Lync follow the classic flow to PSTN (Voice Gateway –> PBX –> PSTN) , it’s important to know that all calls made by Lync can have the same Calling number as the associated extension in PBX dial plan :

– for example if i make a call from PBX phone my external DID is : +3906……4444, but PBX add instead of me the +3906….. (* maybe that +39 is not considered in a national call) .

When i make a call from lync if i want that it must be the same calling number as the PBX phone ,  i have to configure on Voice Gateway a good format for PBX to accept DID so for example in ALCATEL enviroment i must pass to it the call in this format :

calling number (Lync side) +3906…..4444  — >  Translated by VG in  : 06…..4444 , in this way ALCATEL recognize the call as one from its dial-plan, otherwise can appean that my calling number is only +3906……. without the extension.

Result : the call appear to PSTN exactly from one number shared by Lync and PBX and we can realize a true Single Number Reach

Requirement for this scenario

We have to consider that if we make a QSIG trunk between PBX and Voice Gateway my advice is to use a QSIG-GF (Generic Function) not basic because there are a lot of service such as call diversion, line identification, etc.. that is not implemented on QSIG-BC (Basic Call).

If we choose a SIP trunk between PBX and Voice Gateway we have to consider in Voice Gateway side licenses for IPtoIP and eventually transcoding with DSP onboard because if we configure trunk from/to PBX in a RTP codec different from G711, for example G729 , all calls are trascoded (Lync Mediation server work only in G711).

In part 2 we’ll consider a scenario in which I’ll describe the positioning of Voice Gateway in passthrough between PSTN and PBX to prepare a clean migration phase.

Source: http://msucblog.wordpress.com/2012/12/09/pbx-replacement-with-ms-lync-with-dual-forking-part-1/

Just How Could Virtual PBX Boost A Firm

5 Dec

Business VOIP 101

Firms of any size realize the numerous features about virtual PBX or simply often called ‘hosted PBX’ or ‘virtual PBX’. The system arranges a more reliable image for starting off businesses and business people and it also enables them to immediately build up its business opportunities. Entrepreneurs originated the system to be the revolutionized business telephone system in this age. The setup are able to link up and filter phone calls, faxes and voicemails to one multiple telephone numbers. It can be included with MS Outlook. There are plenty of benefits the business can make use of in making use of a hosted PBX. It can potentially be a powerful tool which will help your enterprise take advantage of its success via the advantages. What exactly are those advantages?

• Unified messaging

At this time, there are several methods to send a message. You may transmit it by using a…

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