asterCC released version 0.21 beta

asterCRM 0.075 in asterCC 0.21 beta:

added customer_lead and note_leadadded defined code for note

added user online statistic and user online report

added user type and user privileges

added attend transfer for agent modeadded hangupcause, hangupcausetxt and dialstatus to cdr tablefixed bug of astercc daemon works with asterisk 1.2fixed bug of incorrect didnumber on agent modefieed bug of incorrect curcdr works with astercrm.agiadded doubleCheckCampaign in astercc.confadded export data to daillist directlyadded count and time of tranfer to cdr and dialedlist

asterBilling 0.15in asterCC 0.21 beta:

added pushcall mode(works with pushcall.agi)added billing by accountcode modeadded query and hangup current channels in system page

Related posts:

asterCC 0.14-beta and asterCC BOX 0.14-beta-rc1 releasedasterCC 0.14-beta and asterCC BOX 0.14-beta-rc1 releasedasterCC and asterCC-BOX released version 0.2asterCC and asterCC-BOX released version 0.2astercc 0.2 beta released

tutorial: use astercc,freepbx and asterisk to build a broadcasting system with IVR and Queue agents

the most feature in astercc is the predictive dialer, using the dialer you could improve the work efficiency. In this tutorial, I will introduce how to setup a dialer with a pre-configed IVR:  when dialer start work, customers will hear a IVR which you configed in Freepbx, so we could also config in the IVR to accept customer input, and we can forward to an new IVR or agents in a queue.

1. config a Queue in freepbx

usually you want to config a queue to resonpse customer if they want to reach some live agent, so we config a queue first

2. config an IVR in freepbx

2.1 first we will add some voice in the IVR, you’d like to use a recording software, just notice that in asterisk, it requires to use wav format and 16bit, 8000HZ mono

2.2 add a recording

you can either upload the recording you finished in your pc, or use a ext. in system to record a new one

2.3 add IVR

now queue and recording is ready, we could add the IVR

in announcement, we select the recording we just done, and also we added two options to accept customer input, when customer hits 1, he will go queue 900, and when hits 2, he will go to ext. 5001.

4. add misc

then we need add a misc so the dialer could reach the IVR

remember this feature code 800, we will use it when configing campaign in astercc, you could use some other code as you like, just make sure it’s unique in your freepbx

now we finished the job in freepbx, you’d like to dial 800 from any extension, it supposes to bring you to the IVR.

5. config the campaign in astercc

login as admin, and go to campaign, add a campaign as following

make sure in “Inexten” you put the same code as in “misc application”, and in “Queue number” put the same queue number as your freepbx queue

6. add agent user in astercrm

you’d like to add some account for your agents if you want them get a popup form when they start answer customer calls, go to “Extension” to add astercrm account for your agents, make sure the account “extension” match the ext. in your freepbx

7. start dialer and test agent

before start dialer, you’d like to check your astercc.conf

make sure this parameter is configed as

doublecheckcampaign = yes

then restart astercc daemons

/opt/asterisk/scripts/astercc/asterccd restart

ask your agent login and as admin go to dialer page, start the dialer

as an agent, he will get a popup when customer answered the call and hit 1 for queue

8. check report

you could go to dialedlist to check the campaign status

* to get a working freepbx and asterisk easily, you could choose  astercc box, elastix, pbxinaflash or trixbox

Related posts:

dialer, queue and popup for asterisk callcenter(freepbx,trixbox,elastix,pbxinflash) with asterCRMdialer, queue and popup for asterisk callcenter(freepbx,trixbox,elastix,pbxinflash) with asterCRMtutorial: use astercrm & asterisk for broadcastingtutorial: use astercrm & asterisk for broadcastingbuilding a virtual office using astercrm ,freepbx and asterisk

Login to asterisk queue as dynamic agent in astercrm agent interface

In astercrm version 0.7 ,  we can login to queue which assigned to campaigns as dynamic agent in portal page.

Set “queue number” and “queue context” for campaign:

queue context: which context will be used when login to queue, for example:exteinsion is 8000,set queue context to from-queue,then login to queue as Local/8000@from-queue/n ?

login to astercrm,the queues will display in “Campaign Pannel(Queue)”:

click login button after campaign-1(1000),will login to queue 1000 as Local/8000@from-queue/n, login button turn to logoff:

execute queue show 1000 in asterisk Cli:

then could use pause/continue to control agent if receive incoming call from queue:

Related posts:

cant login queue via agent interfaceDynamic Agent mode with asterisk, FreePBX and asterCRMdialer, queue and popup for asterisk callcenter(freepbx,trixbox,elastix,pbxinflash) with asterCRMbuilding a virtual office using astercrm ,freepbx and asterisktutorial: use astercrm & asterisk for broadcasting

Installing vzdump on CentOS 5.4 (32-bit)

There are a few items required for installing vzdump for OpenVZ on 32bit CentOS.

First, make sure you read the installation post covering the 64bit install. Here are the commands for 32bit. Here is how you get vzdump on a clean version of 32bit CentOS (via the hostnode):

rpm -ivh "ftp://ftp.pbone.net/mirror/ftp.freshrpms.net/pub/freshrpms/pub/dag/redhat/el5/en/i386/RPMS.dag/cstream-2.7.4-3.el5.rf.i386.rpm"wget http://dag.wieers.com/rpm/packages/perl-LockFile-Simple/perl-LockFile-Simple-0.206-1.el5.rf.noarch.rpmrpm -ivh perl-LockFile-Simple-0.206-1.el5.rf.noarch.rpm/bin/rm perl-LockFile-Simple-0.206-1.el5.rf.noarch.rpmrpm -ivh "http://chrisschuld.com/centos54/vzdump-1.2-6.noarch.rpm"

Make sure you add in the PERL5LIB in your export (see the 64bit post):

export PERL5LIB=/usr/share/perl5/

Asterisk 1.8.2.2 Now Available (Security Release)

Asterisk 1.8.2.2 Now Available (Security Release)

Thu, 20 Jan 2011 21:35:00 -0400 The Asterisk Development Team has announced a release for the security issue described in AST-2011-001.

Due to a failed merge, Asterisk 1.8.2.1 which should have included the security fix did not. Asterisk 1.8.2.2 contains the the changes which should have been included in Asterisk 1.8.2.1.

This releases is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2, 1.8.1.2, and 1.8.2.2 resolve an issue when forming an outgoing SIP request while in pedantic mode, which can cause a stack buffer to be made to overflow if supplied with carefully crafted caller ID information. The issue and resolution are described in the AST-2011-001 security advisory.

For more information about the details of this vulnerability, please read the security advisory AST-2011-001, which was released at the same time as this announcement.

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.2

Security advisory AST-2011-001 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-001.pdf

Thank you for your continued support of Asterisk!

Asterisk 1.10 Update

Asterisk 1.10 Update
Greetings,

Shortly after the release of Asterisk 1.8, we had a developer meeting and discussed some of the projects that people would like to see in Asterisk 1.10. We discussed the schedule there a bit, as well. Now that Asterisk 1.8 has settled down and we are well into the development cycle for Asterisk 1.10, it is a good time to revisit the plans for the next release.

At Digium, the biggest thing we have been working on for 1.10 so far is replacing the media infrastructure in Asterisk. Most of the critical and invasive plumbing work is done and has been merged into trunk. Next we're looking at building up some features on top of that, such as adding more codecs, enhancing ConfBridge() to support additional sampling rates (HD conferencing), adding features that exist in MeetMe() but not Co"> .msg_list {margin: 0px;padding: 0px;width: 100%; } .msg_head {padding: 5px 10px;cursor: pointer;position: relative;background-color:#f4f4ff;border: 1px solid #ccc;margin:0px; } .msg_body {padding: 5px 10px 15px;background-color:#F4F4F8; display: none;border: 1px solid #ccc;border-top: 0px; } .msg_body_2 {padding: 5px 10px 15px;background-color:#F4F4F8; }
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Asterisk 1.10 Update

Wed, 16 Feb 2011 18:23:18 -0400 Russell Bryant has posted some information about Asterisk 1.10:

Greetings,

Shortly after the release of Asterisk 1.8, we had a developer meeting and discussed some of the projects that people would like to see in Asterisk 1.10. We discussed the schedule there a bit, as well. Now that Asterisk 1.8 has settled down and we are well into the development cycle for Asterisk 1.10, it is a good time to revisit the plans for the next release.

At Digium, the biggest thing we have been working on for 1.10 so far is replacing the media infrastructure in Asterisk. Most of the critical and invasive plumbing work is done and has been merged into trunk. Next we're looking at building up some features on top of that, such as adding more codecs, enhancing ConfBridge() to support additional sampling rates (HD conferencing), adding features that exist in MeetMe() but not ConfBridge(), and enhancing codec negotiation.

Of course, many others have been working on new developments as well. I would encourage you to respond if you'd like to provide an update on some new things that you're working on.

We would like to release Asterisk 1.10 roughly a year after Asterisk 1.8. This will be a standard release, not LTS. To have the release out in the October time frame, we need to branch off 1.10 (feature freeze) at the end of June. At that point we will begin the beta and RC process. If you're working on new development projects that you would like to get into Asterisk 1.10, please keep this timeline in mind.

As always, comments and questions are welcome.

Thanks,

--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Original Content (C) 2004-2010Matt Riddell
 

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Asterisk: The Definitive Guide book available for pre order and public review
October 21, 2010 Average Vote: 10
The 3rd edition of the Asterisk book is on its way and available for public review and pre-order.

Article number 2500
January 25, 2011 Average Vote: 10
This article marks number 2500. We have been producing articles on Asterisk and related products now since 2004.

Submit your own stories
September 21, 2010 Average Vote: 10
We have made a pretty big update to the Daily Asterisk News. You can now submit your own stories! There is now a link at the top of the Daily Asterisk News. Just click the link, fill out the form and we will check over your story and submit it.

Using locked PAP2 and PAP2-NA with Asterisk
August 23, 2005 Average Vote: 10
VoIPHacker has posted details on how to unlock a PAP2 and PAP2-NA.

Asterisk 1.10 Update
February 17, 2011 Average Vote: 10
Russell Bryant has posted some information about Asterisk 1.10.

Asterisk-Fax
May 19, 2005 Average Vote: 10
Just thought I'd provide a pointer to the Asterisk-Fax site (it came up on Asterisk-Users).

Tropo Now Speaks Asterisk Gateway Interface (AGI)
October 4, 2010 Average Vote: 10
Jason Goeke has posted a link to an article about Tropo support for Asterisk via AGI.

The Everything Asterisk Video Collection
August 5, 2010 Average Vote: 10
Steven Sokol has posted a blog entry on Asterisk Video Resources.

AstriDevCon: October 29th, Washington DC
August 23, 2010 Average Vote: 10
John Todd has posted a note about the AstriDevCon conference which occurs within the Astricon conference.

Asterisk and Kamailio (openser) realtime integration
August 5, 2010 Average Vote: 9.9
Daniel-Constantin Mierla posted a writeup on combining Asterisk and Kamailio.

Interview with Mark Spencer
November 26, 2004 Average Vote: 9.9
We have managed to get an interview with Mark Spencer AKA Markster. Mark Spencer is the creator of Asterisk and by far the most active developer.

ZORG new C++ and Java ZRTP implementation public release
January 13, 2011 Average Vote: 9.9
Andrea Cristofanini from PrivateWave has sent us across a press release about a new Open Source encryption implementation.

Asterisk IPv6 update
February 1, 2010 Average Vote: 9.8
Olle has posted an update on IPV6 in Asterisk and a link to a blog post of his.

Proposal for T.38 transparent gateway design in Asterisk
April 29, 2010 Average Vote: 9.8
Kevin Fleming has posted a proposed design for a transparent T.38 gateway for Asterisk:

Back to life
July 21, 2010 Average Vote: 9.8
Hey all - I am back online after some pretty big projects which have taken all my time. Will be updating the Asterisk news over the next few days.
Asterisk 1.10 Update
February 17, 2011
Russell Bryant has posted some information about Asterisk 1.10.

AstLinux: Beta images for 0.7.6
January 25, 2011
Darrick Hartman has posted details of the latest beta images for AstLinux 0.7.6 - the embedded Asterisk distribution.

Article number 2500
January 25, 2011
This article marks number 2500. We have been producing articles on Asterisk and related products now since 2004.

Crystal Provisioning
January 24, 2011
Crystal Provisioning - The best and the easiest way to deploy, configure and manage any sip phones in an Asterisk environment

People Tech CRM
January 24, 2011
PeopleTech have sent details of their CRM package.

Asterisk 1.8.2.2 Now Available (Security Release)
January 21, 2011
The Asterisk Development Team has announced a release for the security issue described in AST-2011-001.

Safi Communications Suite 1.5.0 Beta
January 20, 2011
Safi Systems are happy to announce the 1.5.0 Beta release of SafiWorkshop and SafiServer, now collectively referred to as the Safi Communications Suite (or SCS for short).

AST-2011-001: Stack buffer overflow in SIP channel driver
January 20, 2011
When forming an outgoing SIP request while in pedantic mode, a stack buffer can be made to overflow if supplied with carefully crafted caller ID information.

Weekly Asterisk Developer Conference Call
January 20, 2011
Bryan M. Johns has posted details about a new weekly Asterisk developer call.

OrderlyQ - The Answer to Call Center Queuing
January 19, 2011
OrderlyQ have sent through a note about their Call Center Queue Solution.

Asterisk IAX clients

AG-168V IAXy by Digium S100-FX by X100P Virbiage 3010: Available direct from Virbiage, or rebranded (e.g. this Freshtel ATA)

American Distributor for ATCOM AG168V Gateway: http://www.iareaphone.com/ Australian ATCOM distributor: http://www.ob-wan.com/voip European reseller of ATCOM AT-320 and ATCOM AG-168V: Upnet Taide European reseller: for all ATCOM products. IP Phones, ATA's and USB phones German / European reseller: http://www.sipmax.de for all ATCOM products. AT 320 PD - SIP/IAX-Phone, AU 100 - USB-IAX-Phone, AG168V - SIP/IAX-ATA Libraries Corraleta Technology SDK Programmable web-based IAX softphone (DHTML/Java/JavaScript) IAXClient: IAX2 softphone library IaxClient SDK for building Cross Browser, Click2Call, WebPhones and SoftPhones (free for non-commercial) Java IAX Implementation, Corraleta Technology SDK GPL'ed version Pure Java IAX Implementation, GPL'ed Corraleta SDK http://callino.cc/jiaxcapplet/: Java Wrapper for the IAXClient library, including applet which does provide javascript support (Click2Dial) libiax2 : Digium's base IAX2 protocol library Public sector training for building Cross Browser, Click2Call, WebPhones and SoftPhones (free for non-commercial) Njiax IAX2 Java library IAX2 library developed in Java by Nomasystems. GNU Lesser Public License. "100% pure Java". A good example on how using this library can be found in this click to talk (with DTMF support): http://www.nomasystems.com/contacto.yaws SimpleIAX for Delphi is a component library for Delphi and C++Builder, allowing to build the VoIP applications. SimpleIAX is built on top of the IAXClient. Applications (see also VOIP Phones ) http://www.messagingbay.com/ IAX2 client for Android/BlackBerry/Web browsers. Search "MPhoneGG" on Android Market or email support@messagingbay.com for more info12/02/2011. Coccinella Multi Platform XMPP client with whiteboard. Since Version 0.96.19 with enabled IAX2 phone integrated. Corraleta Connect Crossplatform click to talk via IAX (Not free) Cubix for Windows (Available for OEM licensing) DIAX for Windows (Freeware for personal use) Ephone for Windows (May be used Commercially free of charge) GnoPhone for Linux IAX Phone for Windows IAX2 Click2Call Cross Browser, Click2Call (free for non-commercial) (Windows) IAX2 WebPhone build with IAXClientSDK for IE. (Windows) IAXComm for Windows, Linux & MacOSX iaxLite for Windows IAXOCX -a free IAX ActiveX Control for Windows. JackenIAX for MacOSX JIAXclient - JAVA applet and library Kiax for Linux and Windows LoudHush for MacOSX OctoPhone -a free IAX softphone for Windows. SFLphone Open source (GPLv3) IAX2/SIP compatible softphone for Linux Voix Manager for Windows Linux and Mac OSX, a switchboard application with an IAX/IAX2 phone integrated Voix Phone Free IAX/IAX2 Softphone for Windows, Linux & MacOSX Yate for Windows, Linux and BSD - an IAX client and server (license GPL). Zoiper - formerly Idefisk for Windows, Linux and Mac
;This text is for the iax.conf file
; Inter-Asterisk eXchange driver definition
;Register commands to connect to another IAX server has to
;be in the [general] section of iax.conf.
;To set up IAX in trunk mode, add "trunk=yes" to your definition in iax.conf.
;IAX Trunking needs support of a hardware timer. See Asterisk timer for more information.
;Do not use both "host=dynamic" and "defaultip=111.222.333.444",
; make sure it is just one or the other (see bugs 558 and 673).
;
; General settings, like port number to bind to, and
; an option address (the default is to bind to all
; local addresses).
;
[general]
port=5036
iaxcompat=yes
disallow=all
;
; All of the following codecs are supported by VoicePulse Connect!
;
allow=ulaw
allow=ilbc
allow=gsm
;allow=adpcm
;allow=alaw
;allow=g723.1
;allow=g729
jitterbuffer=no
[voicepulse]
type=friend
context=dialout
username=
secret=
host= gwiaxt01.voicepulse.com

[IareaNet.Net]
type=friend
auth=md5
username=
secret=
context=dialout
host=208.111.0.45
qualify=yes
jitterbuffer=no

[anotherprovider]
type=friend
auth=md5
username=
secret=
context=dialout
host=IAX2.anotherprovider.com
qualify=yes
jitterbuffer=yes

[8101]test
type=friend
username=8101
secret=
context=dialout
host=dynamic
callerid=Joe Useri<8101> Asterisk phones
VOIP Phones Go back to IAX

Asterisk hardware recommendations

This page and it's siblings suggest some configurations of hardware and software for setting up an Asterisk based PBX. Most of these are based on running configurations in various organisations today. Scaling an Asterisk system isn't easy, there are not simple answers. Next to the samples below we suggest you read about capacity considerations when planning and dimensioning an Asterisk system.
There are a number of VoIP providers that run Asterisk offering IAX connectivity (often with the SIP proxy SER in front of it) . FreeWorldDialup (FWD) and nikotel.com for example employ Asterisk for PBX functions like voicemail, conferencing etc. Quoting Rich Tehrani:
"Currently you�ll find Asterisk (or * as it is referred to frequently on message boards) in corporations where the IS staff is enterprising enough to set this type of telephony server up. You will also find it in numerous CLECs that won�t share this fact with anyone as it is a significant competitive advantage."

Quoting Mark Spencer (in Saarbruecken):
(Transcribed, answering a question about known larger deployments)
One of the implementations that we know about 800 Pingtel IP phones. For busy operators these JAVA-based phones are not ideal since they require regular reboots. Also Vonage is using Asterisk and has made a lot of contributions to the project, they are using it for voicemail. There is also a large analog installation to be found at Frys Eletronics, which is a large US retailer.

Quoting Stefen Sokol (April 2004):
There are carriers using Asterisk to terminate thousands of lines. NuFone has a data center with 80 Asterisk servers in place. These installations require a bit more engineering than the typical PBX server, but the system does scale to extremely large systems.

Francois Lambert, Atelka, Inc:
We are running a call center with 4 Asterisk servers (10 T1 + 1 Channel Bank). We have internally 240 agents using IAX Softphone with uLaw codec.
We have 200 inbound agents and 40 outbound agents. We know that our architecture is scalable to go up in the 1000 agents and more.

ACD for Asterisk by Indosoft

Indosoft Inc. offers Q-Suite, an enterprise grade call center software for Asterisk, the leading hybrid VoIP/TDM telephony platform. Q-Suite comes with a unique, highly developed Automatic Call Distribution (ACD) to enable complex, yet highly efficient, call handling and routing. Queues can be setup with Service Priority and Skills to allow efficient use of Agents with appropriate skill sets and skill levels. Skills Based Routing and Queue Prioritization are the backbone of modern ACD, and in Q-Suite offers a unique and highly efficient ACD layer that functions on top of Asterisk. Q-Suite can be deployed for multi-tenant or single tenant installations that are either premise-based or hosted at datacenters, to deliver remote and distributed architecture with ease. Q-Suite for Asterisk provides the power, innovation and flexibility needed for your increasingly complex customer interactions.

Q-Suite ACD for Asterisk allows easy configuration and setup of ACD with sophisticated, yet incredibly easy to manage, tools for administering your IVRs, Queues, Skills and other routing logic. Because of Q-Suite’s robust ACD functionality and innovative administrative interface, defining and fine-tuning the rules for handling, routing and managing customer interactions has never been easier. Q-Suite ACD for Asterisk provides your organization with the greatest degree of flexibility and efficiency in meeting you business needs.

This page focuses on the details of Q-Suite’s ACD features and functions, which have been designed specifically for Asterisk. For a general overview of Q-Suite as a complete call center software, please see the main Indosoft page on this wiki, or view our website at www.indosoft.com.

Q-Suite ACD for Asterisk Features:

• Skills Based Routing (Agent Skill, Queue Skill)• IVR and Automation to Interface to Data Sources• Routing based on Time, Day, Week, Month etc.• Adapting to Different Call Models• Queue Prioritization, Queue Overflow Music On Hold, Service Level, Branching• Multiple DID/DNIS with Text Identification per Inbound Service• Continuous Status update for Calls/Channels• Notify Select or Notify All Queues• Transfers (Live, Assisted and Consultative)• Call Park, Retrieve and Conference• Detailed Cradle to Grave Reporting• CRM and Custom Application Integration with Script Builder• IP Phones, Soft Phones, POTS Phones• Call and Status Notifications• Web and Native Client Agent Interface• Built-in Quality Assurance and Recording• API for Computer Telephony Integration (.NET and Socket)

Q-Suite ACD supports different types of call models and is ideally suited for both high volume call centers where up-front automation and efficiency in handling calls is required, as well as lower volume, specialized call centers where agility and unique responses are needed.

Q-Suite supports Queue Prioritization for service differentiation, Skills Based Routing with agent skill levels for handling calls from skills based queues, as well as Unified Queues for skills association at call level.

Call Routing, Queuing, Agent Notification and Status Information are highly configurable to enable finely tuned handling of the customer interaction. Once a call is answered, Q-Suite ACD supports Blind, Conference and Consultative Transfers, as well as Call Park, Call Retrieve and Call Conference. With highly efficient channel monitors, all participants within the ACD have a very good view of live status to make quick decisions.
Dialplan Builder (IVR Builder)
Q-Suite ACD for Asterisk makes building complex IVRs and Asterisk Dialplans a simple task. An easy-to-use Graphical User Interface (GUI) tool with a What You See Is What You Get (WYSIWYG) editor allows easy-to-use drag and drop functionality when designing robust Dialplans for IVR and call routing in Asterisk.

All the standard IVR and Dialplans for Asterisk functions are available, including prompts that play voice files, caller option menus, DTMF key inputs, decision-making, and branching or call routing logic. Q-Suite also provides full database integrations, Voice Recognition, Voicemail and Queue Overflow for more sophisticated IVR development. Q-Suite’s unique administrative interface has greatly improved the ease of setting up and administering advanced IVRs and Asterisk Dialplans.

Q-Suite delivers advanced queue management to Asterisk. As with other enterprise-grade ACD, Q-Suite first routes incoming calls into Queues, which can be prioritized for service efficiency. While in queue, Q-Suite ACD holds call information, sorts calls based on defined prioritization rules, and then connects calls to the most appropriate agent based on skills or other business rules. Queues can be fully separate from one another, but configured to overflow based on conditions. There are no restrictions on the number of Queues you can setup.

Popular Queue features in Q-Suite include Music On Hold, Periodic Announcements, Queue Skills, Queue Priority, Queue Overflow, Voicemail and other Call Re-Routing options, as well as SLA and Call Abandonment settings.

Q-Suite’s provides enterprise-grade, efficient and highly effective Skills Based Routing with Queue Prioritization for Asterisk. Queue priority is dictated by the business case. Calls into a higher priority queue are handled ahead of calls from other queues. Once within the Queue, calls are handled based on the wait time. Queue skills determine which agents are qualified. Queue skills are defined when you setup your queues, and agents are assigned specific skills with skill levels at the time of setup. When a call comes into a Queue with multiple agents waiting and available to take the call, agent skills with skill levels apply.

To summarize, skills based routing is applied to route calls to the correct queue, skills of the individual agent to qualify all the agents who are available to take the call, and skill level to decide which agent should handle a call. Queue Prioritization is applied when an agent becomes available with multiple calls waiting in multiple queues. The skills determine if the agent is qualified to handle calls but Queue prioritization dictates the order. For an even distribution, the skill levels and the queue priority should be set to the same value so that the longest waiting call or agent is chosen to break the tie.

This feature is often referred to as the “Time of the Day Call Routing” and delivers the ability to change the way you handle calls based on the time of the day, day of the week, week of the month, after hours, holidays and breaks. Q-Suite provides a single, easy-to-use administrative interface for all scheduling management in order to provide you with the most efficient call handling based upon scheduling requirements. Q-Suite ACD supports a variety of architectures and delivers the full advantages of VoIP telephony. With Q-Suite you can setup a premise-based call center or a remote, distributed (hosted) call center, or a combination of both. Regardless of the architecture you chose, Q-Suite opens the door to remote or at-home agent utilization while providing a stable, secure solution to consolidating your call center infrastructure.

Q-Suite provides full support for On-Hook and Off-Hook agents, whether they be premise-based, remote or at-home. Other popular agent features available in Q-Suite include Free-Seating, Hot-Desking, Alert Before Call Hand-Off, and easy call transferring from the agent station.

Options with agent phones go hand in hand with agent location and notification options. Q-Suite ACD allows agent phones to be an extension within the PBX (phone system) of the call center ACD, or they can belong to an external phone system. Internal phones could be IP phones, softphones or regular POTS phones.

Q-Suite ACD can be setup in Fail safe mode with a built-in design to ensure that a single failure does not result in any loss of calls or conversation. Should there be any single point of failure, including software failure within the components of the system, the unique fail-safe call recovery feature within Q-Suite ACD prevents all ongoing calls from being dropped and allows new calls to continue coming. As a result, the two end point phone devices during a conversation will be able to continue the voice conversation after a failure without dropping the connection. Q-Suite delivers an industry leading Script Builder tool that provides powerful script development and management features, which are also managed from a single, easy to use GUI. Popular features of Q-Suite’s Script Builder include Custom Variables (Text, Number, LOV), Text Blocks with Embedded Variables, Post-to-URL, External Links with Command Line Parameters, URLs within an I-Frame or a Separate Window, Static Webpage Support, Conditional Display, Page Branching, Robust Voice Recording Start/Stop options, and quick transfer options. The open architecture system of Q-Suite allows easy integration to other software in the back-end for processing. All these features are readily available to evolved applications through an advanced set of APIs, which are discussed below. Q-Suite ACD can work under the hood, enabling vertical businesses to embed the ACD for Asterisk into its mature vertical domain. It comes with a sophisticated API to allow CTI interaction using .NET and socket interface, as well as an XML interface for database interaction. This creates an opportunity for business systems with legacy telephony to incorporate CTI for Asterisk and embed Q-Suite ACD into their evolved business domain. Because Q-Suite runs on the advanced Asterisk telephony platform, it comes complete with built-in call quality assurance features, including built-in Call Recording options, as well as Call Monitoring options, including Listen to Agents, Whisper, and Barge. Q-Suite offers a complete set of reports and an open database to integrate with other reporting engines. Cradle to Grave Reporting with granular details is available out of the box for all reports, including Agent Performance, ACD Performance, Queue Performance, Dispositions and Wrap-Ups, to name a few. Supervisor Dashboards and Wallboards provide call center administrators with a view of the call center performance broken down by agent, queue and ACD performances. Wallboards allow supervisors a bird-eye view of the floor performance. Since all these screens are Web reports, an across the board view is provided to management. Periodic and Historical Reporting can be scheduled and emailed automatically to any distribution list. Q-Suite delivers all the above-mentioned features in an incredibly flexible and versatile software bundle. Due to its utilization of Asterisk as its underlying telephony platform, Q-Suite delivers benefits inherent to its infrastructure; benefits unachievable from other popular call center software solutions. Stable remote agent support, distributed architecture, utilizing multiple trunks, and seamless integration of VoIP and SIP are just a few benefits inherent to Asterisk and Q-Suite. This unique combination of powerful software and telecommunications technology positions Q-Suite to meet the most demanding of call center requirements.

Updating Ghostscript on CentOS 5.4 (ghostscript 8.70) (the remix)

A while back I wrote an article on updating to ghostscript 8.63 on CentOS 5.2. I received a fair amount of email and comments on the topic ranging from “thanks” to “you don’t know what your talking about!” I thought I would update the original post to include the latest release of ghostscript (8.7) as well as answer and respond to the best question / statement in the comments… “how do you do that and can you post the SRPMS?” The answer is, yes, here is how I do it and “yes” I’ll post the SRPMs (or more appropriately where I get them).

First, you have to have the latest version of the build environment as well as some prerequisites for building GhostScript:

yum --enablerepo remi groupinstall "Development Tools"yum --enablerepo remi install libjpeg-devel libXt-devel libpng-devel gtk2-devel glib2-devel gnutls-devel libxml2-devel libtiff-devel cups-devel libtool jasper-devel

(you will note here I use the REMI repo)

Next, I obtain the SRPM files from rpmfind.net and ghostscript-fonts from the centos repo:

wget ftp://195.220.108.108/linux/fedora/development/source/SRPMS/ghostscript-8.70-3.fc13.src.rpm wget http://mirror.centos.org/centos/5.4/os/SRPMS/ghostscript-fonts-5.50-13.1.1.src.rpm

Now, I build them using RPM (rpmbuild):

rpmbuild --rebuild ghostscript-8.70-3.fc13.src.rpmrpmbuild --rebuild ghostscript-8.70-1.fc10.src.rpm

… and that is how I do it! If you are feeling exceptionally lazy and have some inherent trust for my builds in x86_64… here they are:

rpm -Uvh http://chrisschuld.com/centos54/ghostscript-8.70-1.x86_64.rpm http://chrisschuld.com/centos54/ghostscript-fonts-5.50-13.1.1.noarch.rpm http://chrisschuld.com/centos54/ghostscript-gtk-8.70-1.x86_64.rpm

Installing vzdump for OpenVZ on CentOS

There are a few items required for installing vzdump for OpenVZ on CentOS.

First, you’ll need an MTA – I suggest making sure you have postfix installed; if you have postfix installed the initial RPM requirement for “MTA” will be handled for you. Next, you’ll need cstream. This installation is slightly more tricky because (as far as I know) there is no real way to gain this from yum unless you use the DAG Wieers repo. Also, depending on what you have already installed you will likely need the Simple Locking file I/O library for Perl.

Here is how you get vzdump on a clean version of CentOS (via the hostnode):

rpm -ivh "ftp://ftp.pbone.net/mirror/ftp.freshrpms.net/pub/freshrpms/pub/dag/redhat/el5/en/x86_64/RPMS.dag/cstream-2.7.4-3.el5.rf.x86_64.rpm"wget http://dag.wieers.com/rpm/packages/perl-LockFile-Simple/perl-LockFile-Simple-0.206-1.el5.rf.noarch.rpmrpm -ivh perl-LockFile-Simple-0.206-1.el5.rf.noarch.rpm/bin/rm perl-LockFile-Simple-0.206-1.el5.rf.noarch.rpmrpm -ivh "http://chrisschuld.com/centos54/vzdump-1.2-6.noarch.rpm"

Since version 1.2-6 of vzdump the location of the modules is not “automatic” and have found it necessary to export the location of the PVE libraries that vzdump requires via this command:

export PERL5LIB=/usr/share/perl5/

Setting VPS Disk Space with OpenVZ

Disk space can be easily controlled via OpenVZ but I have yet to find anyone to actually explain what the heck to “really” do when you need to add more! Everything I have found about OpenVZ just explains the parameters and never shows you how to do it easily. When I need to adjust disk space on an VPS it is usually when I have someone beating up my ear on the phone or my IM so I needed a fast way to expand the disk without worrying about the details.

There are three parameters in OpenVZ which are directly related to disk usage. They are disk_quota, diskspace and diskinodes. NOTE: there are a lot of other parameters that control and effect the disk but this tutorial will only cover the basics!

The parameter disk_quota is a YES or NO value which disables the file system quotas; if you are not worried about the quotas set it to NO and stop reading. Otherwise; leave it set at YES and continue.

The parameter diskspace is the count of 1K blocks available to the VPS in a soft and hard limit. The hard limit is a stop point similar to filling up a physical disk – when you are out, you are out. The soft limit is when the bean counters get angry and the quotatime timer starts. On a basic installation and VPS setup you will have a 1048576 1K blocks as a soft limit and 1153024 1K blocks as a hard limit. The numbers are not crazy as they are derived from base2. Thus, 1048576 1K blocks is 1GB of disk space. Add an additional 10.2MB to the disk space and you arrive at the 1048576 1K blocks. These are the basic numbers for the basic template that ships with OpenVZ.

The parameter diskinodes is the total number of files, directories and links you can have in the container. Think of them as Post-it® notes and each file, directory and link gets a single note. The default basic number is 200,000 for a soft limit for 1GB of disk space and 220,000 for the hard limit. Normally *nix systems will set aside enough inodes for one inode per 4K disk space block. In the default template for OpenVZ they are setting aside enough inodes for 5.2K blocks. Which I’ll write off as either (a) a magic number or (b) a unique calculation I am not familiar with. Thus, because the 4K block inode count for 1GB of disk space should be 262,144 inodes we’ll use the default template values for our calculations and simply multiply times the number of GB requested.

So…

Now…

The question is how do you adjust them quickly and easily. In this example we are going to work with units of GBs. If you need more granularity you will need to divide it back out to MBs but Gigabytes works great for our needs:

First, we need to define the soft and hard limits, next we apply the updated diskspace numbers and finally set the inode numbers correctly based on the ratio we know from the default template:

Here are the commands (and note below for a quick and easy Perl script):

cid=1324gb=5vzctl set ${cid} --diskspace $((1048576 * ${gb})):$((1153434 * ${gb})) --savevzctl set ${cid} --diskinodes $((200000 * ${gb})):$((220000 * ${gb})) --save#!/usr/bin/perl # display the commands to update an OpenVZ VPS with new disk space requirements# 2009/11/15 - Chris Schuld (chris@chrisschuld.com) use strict; print "Enter VPS CID: "; my $_CID = ; chomp($_CID);print "Enter SOFT Diskspace Limit (ex 10GB):"; my $_SOFT = ; chomp($_SOFT); $_SOFT =~ s/[^0-9]//g;print "Enter HARD Diskspace Limit (ex 11GB):"; my $_HARD = ; chomp($_HARD); $_HARD =~ s/[^0-9]//g;my $_INODE_SOFT = ( 200000 * $_SOFT );my $_INODE_HARD = ( 220000 * $_HARD );print "Run these commands:\n";print "vzctl set $_CID --diskspace ".$_SOFT."G:".$_HARD."G --save\n";print "vzctl set $_CID --diskinodes $_INODE_SOFT:$_INODE_HARD --save\n";

Asterisk Elastix Launches Security Module

As the fastest growing and most popular asterisk based PBX software, Elastix has now launched a security module for the community to help administrators to keep their installations more secured. From the version 2.0.4 and on, Elastix will count with this new module.

This module will be included in the distro, by default, and is part of a policy of the Elastix development department to keep new versions “secure by themselves”, and what will keep at minimum the use of additional infrastructure for security, like firewall equipments.

At first, this module will be launched with 3 components: an embedded firewall, an audit report and a week key detector; however, we expect to improve the functionality of this module in next versions.

“This is just the first version. We have many more ideas in mind for security so that in future we will include more and more functionality to this module” Says Edgar Landivar, CEO of PaloSanto Solutions, the company behind the Elastix product. “Many other IP-PBX products do not have similar tools so Elastix is an example to follow”.

The embedded firewall will block network services in order to expose only what’s necessary. It will also contain a set of suggested security policies that can be applied even for users without deep knowledge of firewalling.

“The firewall was a community suggestion that we decided to include and we are very satisfied with what we are releasing to the market” said Landivar.

On the other hand, the audit report will reveal details of logins to the Elastix web interface. This tool will be very useful for tracking unusual activity, including the internal network, allowing us to find the IP address that originated each access.

Finally the weak key detector is intended to alert the administrator if a password easy to “break” is found.

It is well known by many that some administrators often use predictable keys (also called weak) when creating their telephone extensions. This bad practice is a big problem if you decide to expose the computer to the Internet. With the weak key detector the Elastix server warns of this problem to the administrator and to take immediate corrective action.

The security module was released as a beta on January 6th, 2011 and can be downloaded along with the Elastix 2.0.4-beta version.

asterisk · Edgar Landívar · elastix · Elastix security module · firewall · IPPBX

<< Breaking News: Industry 1st Interrupt Routing Adjustable Asterisk Cards Released

More Than Two Million Downloads of Asterisk in 2010 >>

Asterisk SCF: Make Open Source Cloud Telephony Happen

It’s been two months since Asterisk SCF was announced in AstriCon 2010. It really brings a lot of expectation to the open source community and it’s worth our waiting. Asterisk SCF is still in heavy development and will not be expected to be beta-level software until some time in 2011. But are you ready when it hits the ground? If you would like to start developing on Asterisk SCF, please Visit the Asterisk SCF wiki  to find information on downloading and building the Asterisk SCF source, documentation on the architecture, and see the Roadmap.

Asterisk SCF is a framework that allows developers to create real-time communications applications that include voice, video and text and that meet the demands of a full range of uses, from embedded applications to enterprise and carrier solutions. Asterisk SCF is architected to provide the highest levels of availability, scalability, extensibility, fault-tolerance and performance.

Asterisk SCF will be delivered as a system of distributed components that can be deployed in clusters on a single system or on many systems, transparently. The Asterisk SCF platform will support, as a part of its basic architecture, the full range of real-time IP communications, including video, multi-channel wideband and ultra-wideband audio, chat, desktop sharing and other media types that may arise in the future.

Asterisk SCF is not a replacement for Asterisk, the world’s most widely used open source voice communications platform. Digium and the Asterisk community are committed to the continued development and support of Asterisk, the telecommunications software.

asterisk · Asterisk SCF · AstriCon 2010 · digium · embedded application · open source

<< More Than Two Million Downloads of Asterisk in 2010

The First Open Source Telephony Conference in 2011 >>

More Than Two Million Downloads of Asterisk in 2010

Before turning off my laptop, an exciting new announcement came in my sight: Asterisk had more than 2 million downloads in 2010. That is to say about an average of 5500 downloads per day. And this is the growing power of the world’s most welcome open source PBX software! I would love to share the news with everyone of you.

Digium caps off a year marked by strong growth in the use of Asterisk and substantial technical advances in the product. Contributions from the open source community matched Digium’s investment in Asterisk over the past year. To date, more than 9,800 people have contributed code to Asterisk, including more than 200 who worked on Asterisk 1.8. After releasing version 1.8 in October, the momentum continued later that month when the company also announced a new open source project, Asterisk SCF™. Enthusiasm for Asterisk among users, including developers, resellers, integrators and systems administrators, also increased as they downloaded the software more than two million times in 2010.

Asterisk turns an ordinary computer into a communications server that can power IP PBX systems, VoIP gateways, conference servers and other communication applications. In over 170 countries today, small businesses, large enterprises, call centers, carriers and governments are using Asterisk to create standards-based, feature-rich communications systems at a fraction of the cost of proprietary systems. Digium estimates that more than one million servers around the world are currently running Asterisk to handle billions of minutes of phone calls.

“Asterisk has made an indelible impression on the voice communications industry in the 11 years since it was released,” said Bryan Johns, Digium’s community director. “Its appeal keeps growing as businesses look for the value, flexibility, standards compliance and the technical superiority that result from the contributions of thousands of talented and visionary software developers. Digium is proud to sponsor Asterisk and to be a part of its community, which now counts 73,000 registered members. The company continues to focus on Asterisk’s development, as we saw with the release of Asterisk 1.8 this fall and with the creation of a new open source project in Asterisk SCF.”

Well, enjoy it!

asterisk · asterisk downloads · Asterisk SCF · call centre · conference server · digium · IPPBX · open source · VoIP

<< Community News: Elastix Launches Security Module

Asterisk SCF: Make Open Source Cloud Telephony Happen >>

Asterisk consultants worldwide remote

The following consultants provide remote Asterisk services to customers worldwide. They may also provide onsite services; see the onsite page for details.

Add your entry here (Alphabetical order by company):

Evariste Systems Freevoice Telecom U.S.A. Fortuitous Technologies FluentStream Technologies Generation Software Solutions Godson Gera - Hyderabad - India GreenfieldTech CTS Consulting & Trade Service GmbH HomeUnix HireRussians @ sibers INTA Integrics Ltd Chennai, India People Tech systems private Limited INTELLECT Information Technology Pty Ltd Intuitive Voice Technology Jared Smith, Asterisk Consultant IP COMMUNICATION doo iPhonica LLC. GEDAM Europe SRL, Torino hagel IT-Services GmbH, Hamburg ITConnection iTFUTURE Tecnologia e Informação Ltda. (São Paulo-SP / Brasil) Mathias Kettner MikeOverIP Modulo Consulting SRL monoploid Linksoft Technologies Ltd (Hong Kong) Moki Systems Nasoni & Friends NDSoftware Nethawk (Pvt) Ltd NextGenData Next I.X., inc. Su socio tecnológico Oleg Titov, Asterisk Consultant Lintel Technologies Pvt Ltd Pbxnw.com PCDOCTOR - VoIP Division Progressive Voice Services Limited http://www.raynettech.com/ SineApps SIP:XX Communications - "Double-Xtra-Strength SIP!" SMB Consulting www.stuartelvish.com SviluppoIT The Sjobeck Company LLC Synaptic Pty Ltd The Uptime Group, Inc. Ulterius Pty Ltd VoIP Reach Communications Voiptech, Italy Voiscout, Germany, Hamburg Warrior Hill, LLC Italy

ASTNEXT SOLUTIONS PVT LTD
ASTERISK CONSULTANTS(GUJARAT,BANGALORE,INDIA)

AstNext was founded with intention to market Open source culture which is the creative

practice of appropriation
and free sharing of found and created content.

AstNext is able to provide expert guidance and technical support as we are focused on Open

Source IP PBX based telephony solutions.
Our affordable options can reduce monthly phone expenses,
offer you the flexibility to work from home or create a professional phone network in your office,
or even allow you to connect multiple offices with one system and a common set of phone

numbers and extensions.
Our team is dedicated to high-quality service.
We work to ensure that your applications have the greatest impact for your target audience.

We are Asterisk service provider, we are leaders in providing turnkey solutions in Asterisk

  * Asterisk: IVR/IP PBX/SOFT SWITCH 
  * Asterisk Gateway Interface (AGI) 
  * End to End VOIP Set-up 
  * VOIP Minutes 
  * Predictive Dialer 
  * Call Centre Solutions 
  * Hospitality Telecom solutions. 
  * Medical Telecom solutions. 
  * ACD for PRI/Analog/GSM Phones 
  * Softphone 
  * SER/billing soft switch 
  * LCR/CDR 
  * Web to call 
  * Click to call 
  * Inter-office/Intra-office Intercom 
  * IP Phones 
  * IP PBX,Installation,maintenance,configuration of linux systems,servers anddevices. 
  * Support for digium / sangoma / rhino E1 / PRI / FXO Digital / analog Telephony Cards 

/grandstream
Regards,
AstNext Team
Email: info@astnext.com
www.astnext.com
US: 209-349-9648
Cell: +91 8000770962

Having experience in asterisk and VoiP for more then 5 years
setup loadbalanced asterisk for more then 2000 users
setup many asterisk installation for Companies, call shops and contact centers
have done many Small to medium single and distributed load balanced Vicidial Installations
Of more than 300+, Seats plan done.
Providing support to 25 call centers at different locations in India
Provide remote and onsite Support with turnkey solutions
Provides installation and support for:
Asterisk Asterisk + vicidial Asterisk +video (GNU gk/Open MCU) gatekeeprs SER fax solutions fax broadcasting Voice Brooadcasting IVR solutions Auto Dialers sip gateway A2billing asterisk + trixbox gnu dialer Asterisk integration
Also provide support for existing vicidial, asterisk and trixbox setups any were in world
And do the configuration both onsite and remote configuration
Contact: aabhas bajpai
E-mail : aabhas.bajpai@gmail.com
Phone : +919160439711,+919936283432,
web : http://voip-info.org/users/view/abhasbajpai
India.
aabhas bajpai Having Experiance in Asterisk and Voip for more than 2 years
succesfully setup 200+ seats contact centre for various Activities(Sales, Collection etc)
Provides Remote and onsite support
Provides installation and support for:
   *  Asterisk Installation
   * Asterisk plus vicidial Installation
   * Asterisk integration
   * Elastix
   * Auto Dialers
   * Vicidial
   * Voice Broadcast
   * Voice/Video conferencing setup 

Also provide support for existing vicidial, asterisk setups anywhere in the world
And do the configuration both onsite and remote configuration
Contact: Ajay kumar
E-mail : aty2570@gmail.com
Phone : +919869559250
Web : http://www.voip-info.org/users/view/aty2570
India.

Web site) http://www.vontrade.com Phone) 416-884-2282, E-mail: ali@vontrade.com URL) sip:99999@it.vontrade.com Asterisk installation and configuration locally or remotely on Windows or Linux Pre-paid and Post-paid, Billing, Call rating, Least Cost Routing Voicemail, Fax to mail, push to talk, SMS, DID, IP phone provisioning ASR, ACD, PDD, Interactive call monitoring and online call control Database driven IVR and Configuration, web portal configuration SIP development specifically Soft phone. Script programming, AGI Development, Perl and PHP Turnkey VOIP solution for Retail and wholesale business
Based in Oman and located in more than 10 centres all over the middle east and possess more than 37 years of industry experience.
Services offered are:
Installation of Asterisk PBX with PSTN, GSM and FAX integration Installation of switches with trafffic management and Wifi Networks with wireless IP phones Turnkey solutions & Integration of asterisk with existing network Call center solutions Billing and CRM solutions Structured cabling and supply of all passive and active networking components including PC's and peripherals.
Phone:+968 24816303
Email: it@alkhalili.com
Web: www.alkhalili.com
http://www.asteriasgi.com sales@asterisksgi.com Phone Toll-free: 877-ASGi-4-ME (877-274-4462) ... or Local in Huntsville, AL: 256-705-0277 Digium | Asterisk Premier Solution Provider and Partner Design, Configuration, and Installation of Asterisk PBX Systems Deployment, Configuration, and 24-Hour support of our OmegaDial(tm) auto-dialer platform Integration Asterisk Consulting is our Specialty! Our products include: A Call Center Agent GUI, an Asterisk Queue Management Software, an Asterisk Autodialer, and a Graphical Asterisk Configurator and IVR Builder Flexible Payment Scheduling to meet your financial needs. We are based in Huntsville, AL, but service the entire World! Call today to find out what we can do for you!
http://www.AsteriskAgent.com/ Worldwide Support support@AsteriskAgent.com 800-763-2908 Specializing in Asterisk Based solutions. Asterisk Support & Staffing Live Website Support Available
Asterisk@Office is a leader in software applications and system used for IP-Telephony. Our consulting services help your organization connect its existing telephone lines with a state-of-the-art hybrid VOIP environment where inter-office, inter-branch and customer calls can take advantage of Web based voicemail, Conferencing, Automatic Call Distribution and many other advanced features.
Asterisk Design, Installation and Maintenance Vicidial,a2billing,Click2call nad many more Failover solutions , interconnecting multiple Asterisk servers Installation and Configuration support on asterisk,vicidial,a2billing,openser, We work on Hourly basis
Most affordable Price !! Visit us at http://www.aasteriskneeds.com
Highly professional support for home and office PBX systems based on asterisk. Expertise in Trixbox, Asterisk, FreePBX, Hylafax, ATA, OpenSER, FreeRadius Asterisk module development and source code customizations Custom application development based on asterisk IVR systems Billing Systems OpenSource Consulting Fax system
E-mail :info@asteriskware.com
web : http://www.asteriskware.com Contact us to discuss requirements We offer various services for Asterisk such as Asterisk installation, Asterisk configuration, Dialplan programming, IVR Designing, AGI scripting, manager api, Customized Asterisk Development, text to speech, speech to text, etc.

+1 888 505 VoIP | +1 212 937 8855 Asterisk Consulting Computer Telephony Integration Custom IVR's Advanced Asterisk Applications Trix Box Support Asterisk Support PBX Planning & Network Design VOIP Phones and Hardware Linux Administraton VOIP Server Hosting and Colocation
Website:http://www.astravoice.com
Phone: +1 212 937 8855 - +1 888 505 VoIP
E-Mail:info@astravoice.com We provide services for your all Asterisk projects, installation,
integration, custom development and online Asterisk support.

In English, En Français, Türkçe

Provides installation and support for:
Asterisk Asterisk + vicidial IPBX configuration 1 to 5000 extensions OpenSER sip gateway A2billing asterisk + trixbox ViciDial (Call Center) Asterisk integration Calling Card (open source) Call Shop (open source)
Contact: Ahmet Yilmaz
E-mail :info@atyconsulting.com
Phone : +33(0) 954 974134
web : http://www.atyconsulting.com
France
Ahmet Yilmaz We are specialized on Asterisk, and GNU/Linux, we offer consulting, remote assistance, monitoring and personalizations.
Feel free to contact us.
         o Location: Padua, Italy
         o Tel.: +39 049 796 72 72
         o E-Mail: info@beffect.net Asterisk Consultation Asterisk / Windows Network Intergration Turn Key Asterisk VOIP Systems VOIP Phones and Hardware Linux Administraton VOIP Telephone Service Provider National Internet Provider VOIP Server Hosting and Colocation Hosted VOIP Provder
Website:http://www.BIGNetworks.com
Phone: +1 225 214 5355
E-Mail:voip@bignetworks.com We not only install complete asterisk systems we also integrates customs solutions for each business with our vast experience
Working with asterisk and programming languages like pearl and .NET allow us to accomplish any need you may have
We go to a simple asterisk installation or web page until solutions like our own predictive dialer, CTI platforms, CRM integration
And Telco support.

We offer remote and on side support worldwide.

AgileCO.net offers Advanced Integration, Development and implementation for OpenSER/Asterisk/OpenWengo. Our professional services include the following:
SIP, Skype, GoogleTalk, 3GP-IMS, P2P-SIP, VoiceXML, Mobicents; Mobile-SMS, J2ME, ParlaX and XML WebServices SOAi.

Operate in Europe; Asia and Americas
   * Web site: http://agileco.net
   * E-mail: info at agileco.net
   * Phone: +16789357099
   * skype: edpimentl
   * Professional Services
         * OpenSER/Asterisk Installations
         * SIP/Skype/Mobile-SMS/J2ME Software Development
         * Project Management
         * Project Outsourcing 
   * IVR/VoiceXML Development
   * Custom AGI Development
   * Support Contracts 

Considering deployment of an Asterisk solution? We can provide design, installation, configuration, maintenance and support of single and multi-site configurations. We bring experience of consumer retail solutions such as calling card and low cost call solutions, call centre deployments with CRM integration, and integration with low cost VOIP suppliers.

Our most recent project is a low cost conference call service for the UK: http://www.ConferenceGenie.co.uk

Providing Experienced Asterisk deployment and customization. Small to large scale implementations including integration into enterprise Java, .NET, Databases and more. CallFire provides highly-scalable and affordable Cloud Telephony and Homesourcing solutions, including Voice API ,Voice Broadcast , Virtual Call Center , Hosted IVR , Call Tracking, Phone API and VoiceXML services.


Bitnetix Technology Consulting is a privately owned business dedicated to providing expert advice in all areas of technology, IT, computers, and telephony to credit unions, not-for-profit organizations, and small businesses. Located in Western New York, we offer a wide range of services to local clients as well as providing off-site assistance and travel to remote locations and other geographic areas.
VOIP Asterisk FreePBX Lots more Visit www.bitnetix.com for information on how to contact us, or give us a call at 1.877.33.VOICE (877-338-6423) Web site: http://callforsolutions.com/ E-mail: info AT callforsolutions dot COM Phone: +1 (514) 667 - 0716 Professional Services Asterisk Installations And others services With over a decade of experience in the computer field we can provide a variety of services such as installation, configuration and support of a big variety of products. provides VOIP consulting, installation, and other technology services.
Custom VOIP application and component development & deployments for commerical and residential needs.
Specialties include fault-tolerance, security, IVR, voice-recording, translation, and scalable clustering.
Economy and Premium services available for SIP origination and termination. Multilingual team for support in your language.


Asterisk for C/C++ programming. Very good at monkeying with the voicemail app.
20+ years of professional programming in telecom networking. Asterisk SIP, RADIUS, AAA, RtpProxy, MGCP, and voicemail changes. No project too small. Rates from $65 to $165 per hour (Depending on how good your specification is and how much of the project you want us to control). All work done on fixed price basis. Costs depend on detail in specifications. You get a free lecture with each poor specification.
Voicemail specailist. (if someone came up with the money I would rewrite the damn thing in C++)
All work is handled remotely; believe you me you could not afford to have me on-site...

I am also doing Ruby on Rails with Video over IP for Security Applications now.
I have just completed changes to OpenSER & RTPproxy to record calls to wav files.

"Your Complete VoIP and IP PBX Consultants"
http://www.convertecsolutions.com
Convertec provides businesses of all sizes with expert, professional, and guaranteed Asterisk, PIAF, FreePBX, and Trixbox support and help.
We offer email, telephone, and remote session based assistance by the hour or with blocks available starting at ONLY $65.00/HOUR!
Why not select a VoIP consultant that not only provides support, but also provides SIP origination AND termination services? Convertec Has you covered!

We currently provide the following services
US DID coverage to over 90% of USA ratecenters VoIP and PSTN termination Toll-Free and Vanity Toll-Free numbers eFax services with desktop tools, online portals, or email receipt VoIP Network Support & Management 24x7x365 On-Call Engineer support services Emergency support services payable via credit card are available Asterisk, FreePBX, Trixbox, PIAF troubleshooting, diagnosis, and repair Complete installation support and coaching/training
Info@ConvertecSolutions.com OR 800-928-3109 Option 1 Createlsoft is a services company located in Delhi.We provide telecom solutions to Enterprises and Telcos.
We provide solutions based on popular Open source products (Asterisk, OpenSIPS...).
For any enquiry, please contact SURESH ARORA.
Email : sarora180673@yahoo.co.in
Mobile: +91-9810252984
Skype: suresh_arora73 A-5020 Salzburg, Fürstallergasse 36
D-83395 Freilassing, Sägewerkstrasse 3

Asterisk server consulting, solutions, integration with legacy PBX systems. Pre configured Servers, installation, maintainance. Open Source specialist. Company has a wide range of references thruout Europe. We do FreeBSD, Linux and Windows.

We have built networks for calling cards, conferencing, call centers, outbound dialers, custom IVR development, advanced database integration, embedded Linux product development, automated testing, and softswitch integration. Some of our installations have 1,000 lines or more in service.
Web site: http://www.directleap.com/ Located: Toronto, Canada E-mail: info @ directleap.com Phone: (416) 405-8073 We do high-volume and highly-available carrier-grade Asterisk / SER engineering. We deliver strategy-driven technology for marketing, operations, and product development. We will make your installation carrier-grade, with automated monitoring and testing. Let us give you advice.
Asterisk/VOIP/Telecom Consultant Services offered: Installation/maintenance/configuration of linux systems/servers VOIP Gatekeepers/Phones/devices. Small, Medium, Large Scale deployments of Asterisk, SER, Asterisk@home/Trixbox/FreePBX/Asterisk2Billing/Astbill Asterisk installation/configuration/Customization, Dialplan programming, AGI/perl scripting, Text-Speech/Speech-Text integration, Speech recognition Vicidial installation, customisation and support Inbound/outbound/predictive dialing solution for call centers. Onsite and remote assistance provided, can travel worldwide to implement large projects. Msn messenger: diyanat@hotmail.com Skype: diyanat Phone:+91 98850 39230 Voip: +18473801847 Email:diyanat@hotmail.com __ http://www.voip-info.org/users/view/diyanat
On the Web@ http://www.e4strategies.com
Traverse City IP PBX / VoIP, IP Telephony Consulting Company
Specialization- Traverse City PBX, IP Telephony, VoIP, (CRM) Client Relationship Management, and Carrier Services .e4 makes technology work for your business. We see technology as your engine for growth — helping businesses become more adaptive, competitive and profitable. We partner with industry leaders to deliver the hardware, applications and expertise to advance your business through the intelligent application of technology.
.e4's unique approach combines proven methods such as process management and known industry best practices together with forward-thinking concepts like running IT "as a business" not "within" a business.
By adopting e4's best practices companies are empowered with the ability to deliver quality services that satisfy business needs, are process-driven, meet cost targets, and achieve performance goals.

Contact Us
.e4 LLC
335W South Airport Road
Traverse City, MI 49686

Edwin is located in Manila, Philippines. Has extensive experience in setting up inbound and
outbound call centers. With experience as a Call Center IT Manager, Edwin knows what do you need in regards of call center technologies. Now currently as a consultant Edwin has installed and currently maintains various VICIDIAL implementation
ranging from 20 seats to 300 seats call centers. Edwin is also a member of VICIDIAL Vendor Association.

Services offered:
VICIDIAL Installation / Configuration
Custom VICIDIAL add-ons
Asterisk Installation / Configuration
Call Center Technology Setup
Asterisk billing solutions
Call Center Technology Consulting

Contact: Edwin A. Valbuena Jr.
Email: edvalbuena (at) gmail.com
Phone: +63921-773-1927
Mobile: +63915-789-4984
Website: Core Communications Inc.

Elisha Telecom offers both on-site and remote VoIP/Asterisk Consultancy worldwide. With staff located in the UK and the Philippines, our team is on hand 24/7 regardless of your time zone. No job is too small or too large.

Our expertise was obtained by implementing predominantly Asterisk based solutions in call centers, including the provision of sophisticated "call back" and conferencing solutions. As well as working with end users, we also work with other Telcos to offer support and advice with trouble shooting on an ongoing and ad hoc basis.

Our services include:
   * Asterisk PBX installation, setup and maintenance
   * Call Center setup
   * Asterisk AGI/AMI scripting
   * Custom Web based Asterisk GUI and CTI development
   * Legacy PBX integration
   * Open Source Convergence and Collaboration solutions 
   * White Label Hosted VoIP option
   * VoIP Origination and Termination (Wholesale and Retail)
   * VoIP Hardware (Digium TDM Cards, Polycom IP Phones, and more)
   * VoIP Consultancy and project management

Contact us for an informal chat and see what Elisha Telecom can do for your business:

The largest open source solutions provider in Pakistan. Our wide range of services and supported technologies are:

   * Remote Server administration & management
   * 24/7 Helpdesk support
   * Asterisk Integration
   * ViciDial integration 
   * CRM integration
   * Billing software
   * Trixbox expertise
   * OpenSER
   * FreeSwitch 
   * Voice/Video/Web Conference integration
   * IVR Integration

Contact us

http://www.evtmedia.com/ services "at " evtmedia.com (<-try this first - we travel a lot!) Phone: USA: +1 650 594 4630 , UK: +44 (0)207 043 3018 (NEW) Specialists in High-volume IVR, Police & Security Applications, Speech recognition, and Multi-lingual systems. We also provide micro-coding for specialized Bluetooth transmission software (Proximity Marketing apps) Offices in Silicon Valley, California and London, England Will Travel: Western Europe and USA
Asterisk India - Nurturing Asterisk Community in India - The Authentic Asterisk India Website
Enterux has been the pioneer in the field of Asterisk and Open Source Communications. It was instrumental in starting the asterisk revolution in the Indian market by starting the first ever Indian asterisk users community forum. Over a period of time, Enterux has been very much active in the Government and large enterprise sector and has also provided training to scientists from Depart of Defense - Center for Artificial Intelligence.
Enterux is the only company in India who has Digium Certified Asterisk Professional (dCAP) consultants on board.
Our consulting services help your organization connect its existing telephone lines with a state-of-the-art hybrid VOIP environment where inter-office, inter-branch and customer calls can take advantage of Web based voicemail, Conferencing, Automatic Call Distribution and many other advanced features.

Services Rendered:
Predictive Dialer Medium to Large Multi-Office Business Telephony Systems Call Centers - Local and International Asterisk Dialers and Bulk Calling Systems International Office Telephony Systems Specialty Calling Systems (Entertainment and Personals) Healthcare Application Integration Customer Relationship Management, CRM Application Integration Distributed Server Architecture and Asterisk Load Balancing SIP Express Router, SER Load Balancing Hospitality Telephony Systems (Hotel PBX Integration) Complete IVR Development Local or Datacenter PBX Customization Wireless Telephony Installations Database Integration and Customization Custom Application Development IP PBX/ Installation / maintenance / configuration of linux systems / servers VOIP Gatekeepers / Phones / devices. Distribution & Support for Digium range of hardware in India (E1 PRI / FXS / FXO / Echo cancellation / G729 Convertor Codec & Card) Distribution & Support for Xorcom range of hardware and software products in India (E1 PRI / FXS / FXS / AstriBank / AstriBank Server) Distribution & Support for Sangoma range of hardware and software products in India (E1 PRI/ FXO/ FXS / SS7)
Contact us to discuss requirements customizable pbx turnkey solution - e.com pbx VoIP consultancy and research pbx and soft-switch developement and integration callcentre desing and implementation
epbx Headquarters
Gotthelfstrasse 102, 4054 Basel, Switzerland
phone: +41 615115014
www: http://www.epbx.eu

epbx Czech office
V Haji 1214/13, 170 00 Prague 7, Czech republic
phone: +420 226517511
www: http://www.epbx.cz

Supporting asterisk based telephony systems since 2005, references available upon request. Also providers of SIP/IAX origination and termination services with over 90% of US ratecenters on-network. LNP available. NO contracts required We configure and troubleshoot Trixbox with VoIP, PRI, Digium, Rhino, ZapMicro, Sangoma , and many more hardware providers Guaranteed to fix your dropped calls, echo, "all channels are busy now", and all other common troubles. Assistance with phone configurations, features, and other asterisk/endpoint issues Hourly and contract pricing starting at ONLY $65.00/HOUR!!! We can service anyone anywhere in the world via ssh and web access. 24X7 emergency assistance is always available. 800-928-3109 | 208-664-2869 | www.convertecsolutions.com | info@convertecsolutions.com
Web site: http://www.evaristesys.com/ Knowledge base: http://www.evaristesys.com/workshop VoIP FAQ: http://www.evaristesys.com/workshop/index.php/VoIP_FAQ Contact: 678-954-0670 (or see contact form on web site) Location: Atlanta, Georgia, USA Specialisations: Kamailio (OpenSER) support & development (management advisory council member, official ASIPTO partner) Large-scale Asterisk deployments. SIP service delivery platform engineering. IP network engineering. VoIP contact centers project. QoS. VoIP application integration. Telecommunications operational support systems. Billing, rating, & least cost routing. Cisco voice gateways. SIP trunking. Private VoIP peering/settlement. Fax to email. Managed server colocation for VoIP.
http://freevoicetelecom.com/ E-mail: support@freevoicetelecom.com USA: 1-866-907-6158 Asterisk, Trixbox, & Freepbx $50 per hour Support, Maintenance, Upgrades, Custom Integration, & Consultation Digium Certified Asterisk Professional Onstaff Web Site Click to Call Integration CRM Integration Custom Asterisk Dialplan & AGI Scripting Several Outstanding references available Integrate Asterisk with existing PBX Systems
Performance Design, & Capacity Planning
http://fortuitous.com/ E-mail: support@fortuitous.com USA: 1-512-351-7783 System Performance Tuning and Debugging Capacity Planning for Enterprise Asterisk High Availability, VoIP Cluster, and Reliability Design Operating System and Kernel Tuning Network Stack Tuning & Optimization SAN, NAS, and Parallel Filesystem Design: GPFS, ZFS, GFS Full Network and Server Integrated Design Integrate and Tune Asterisk systems with SAN
Worldwide customers and partners! Small through Large Enterprise Asterisk designs and installations Highly Available solutions, Clustering, DUNDi Unbiased advice for SIP origination, termination, and network management Advanced monitoring / managed services offering Asterisk and FreeSwitch Virtualization and cloud / hosted design

A division of Oxford Consulting Group
Remote support: international via SSH / RD / PC Anywhere / VNC Local support: Ontario & Detroit Email: gss@ocg.ca Phone: London: (519) 672-8238 x270 Toronto: (416) 848-7506 x270 Detroit: (313) 483-1424 x270 FWD: 652811 WWW: http://www.generationd.com/ Asterisk and VOIP telephony consulting. Expierence with call centers, large office, small office, carriers Expertise integrating Asterisk into mixed Linux / Windows environments Custom AGI scripts, diaplan extensions, billing systems Linux & Windows admin, config, and security expertise (including ISA Server)
Asterisk/VOIP/Telecom Consultant Services offered: Installation/maintenance/configuration of linux systems/servers VOIP Phones/devices. Small, Medium, Large Scale deployments of Asterisk, OpenSIPS,Kamailio, Trixbox/FreePBX/Asterisk2Billing/Astbill Asterisk installation/configuration/Customization, Dialplan programming, AGI/AMI/python ,Text-Speech/Speech-Text integration, Speech recognition. IVR, Click2Call, Indian languages speech recognition, High Availability solutions. Vicidial installation, customisation and support Inbound/outbound/predictive dialing solution for call centers. Onsite and remote assistance provided, can travel worldwide to implement large projects. Msn messenger: godsong@hotmail.com Gtalk:godson.g@gmail.com Skype: godsongera Phone:+91 9246461828 Voip: +1253242384 Email:godson.g@gmail.com http://godson.in GreenfieldTech is Israel's premier solution provider for Asterisk and Open Source Telephony and VoIP infrastructures. With over 10 years experience in the Open Source IT industry and 5 years experience with Open Source Telephony projects (Asterisk, GnuGK, FreeRadius, OpenSER) - GreenfieldTech is your best choice for Open Source Telephony solutions in Israel.
GreenfieldTech also provides custom platform development services, serving as a one-stop-shop for all your analysis, design, proof-of-concept, development and deployment requirements - with proven track record.

Services offered via remote services:
Custom IVR and telephony application development, based upon Asterisk and Open Source structures. Custom application development for value added services. Integration of Open Source Telephony packages such as A2Billing, FreePBX and others. Integration of Open Source and Closed Source products, achieving best-of-breed architecture and solution. Training services for Asterisk and Linux (Certifications: dCAP, GNU-LCA, GNU-LCP, GNU-LCI).
A-5020 Salzburg, Fürstallergasse 36
D-83395 Freilassing, Sägewerkstrasse 3

Asterisk server consulting, solutions, integration with legacy PBX systems. Pre configured Servers, installation, maintainance. Open Source specialist. Company has a wide range of references thruout Europe. We do FreeBSD, Linux and Windows.

Asterisk PBX installation, configuration and support
provides different Asterisk based solutions with advanced features.
complex configuration, integration, high availability solution and clusters, DUNDI. Using Voice-over-IP (VoIP) technologies we offer you the most cost-effective and reliable way of voice communications via IP networks. We have wide experience in establishing and configuring VoIP systems integrated with such hardware as Cisco, Dlink, and Avaya. We can implement your VoIP systems in the following areas:
Security. If you require establishing security systems with remote access in your office and home, we can provide you these security systems by integrating VoIP with Smart Home system.
Business Automation. We can integrate your VoIP gateway with CRM/Call Centers or ERP systems. You immediately receive all the bells and whistles of digital voice stream such as recording, integration with the database, voice recognition and routing, etc. free.
Communication. We deployed and tuned in a new communication system based on Asterisk Open Source Project. This designed system can be tuned and integrated with your corporate software without the necessity to develop it from the scratch.
* Websites:  
Phones: 1-800-521-4091 (toll free), 1-650-681-4541 (CA) 7 383 3306626 (HireRussians/Sibers Russia) Email about:info@sibers.com
Asterisk integration.
Based in Vilnius, Lithuania. Web site: http://www.inta.lt/ Telephone: +37069811803, +37052120388 Email: Marat at inta d0t lt
Business software integration consultancy. Specialists in Asterisk, SER, Cisco Call Manager, IVRs, and PBX integration, including billing, redundancy, conferencing.
Web site: http://integrics.com/ Telephone: UK 0808 178 3170, USA 1-888-INT-E111, Rest of world +44 20 799 39 799, sip:sales@sip.integrics.com Email: sales (at) integrics (dot) com
Intelegent firmly believes in the power and ability of open source and open standards to deliver feature-rich, readily scalable, innovative and cost-effective solutions to technological challenges. We interact very closely with the open-source communities world-over to constantly innovate, optimize, contribute by sharing and learning to ensure that our clients get the best of what there is.
Asterisk based SOHO / Corporate PBX Converged Communication Solution - EMail + IM + VoIP Call Center (Predictive Dialer) Solutions Interactive voice response (IVR) Solutions
Asterisk Turnkey Solution providers Rate: Starts at $40 per hour Contact: Rajeev Natarajan Telephone: +91 9444072925 / +1-360-519-5969 Email:support AT intelegentnetworks DOT com
IVR,ASR,TTS,VXML,IMS applications, asterisk /Vyatta appliances , SS7 /SIP/TDM Gateways, Sangoma Hardware,Content Security
Contact Center Development , consulting
People Tech Systems Private Limited
F 105,3rd Street, No:4 ,Golden Brook,1st Floor
Anna Nagar East,Chennai -600102
Mob: +91 9789982453 | Dir: 044-26260810 | 044-26220019 Voice Over IP experts, Connecting businesses with VoIP since 2003
Asterisk, Switchvox, Call Manager and other VoIP PBX systems Customer-oriented IT integration and support company Expertise in communications and network systems, not only VoIP! Design, Implementation and Support of VoIP systems
With Intellect IT solution you can do telephony in new ways:
Connecting employees working from home to the office PBX over broadband connections. Enabling remote users to have calls directly transferred to them as if they were working from their office. Giving all employees voicemail, integrated with the Web and their E-mail. Allowing users to retrieve their voicemail messages from a web interface or have them directly delivered to their e-mail. Building interactive voice applications, that connect to your ordering system or other in-house applications Giving access to the company PBX for business travellers, directly from the laptop computers using a soft phone application connecting from airport or hotel wireless hotspots Free calls between interstate and International offices. Free calls to mobiles Virtually unlimited extensions and voicemail ...and much more Contact us to discover how our solutions will meet your business needs
Website: http://www.intellectit.com.au.
Phone: 1300 799 165 (+61 3 83171000) -=Easy to Use Full Featured Asterisk Solutions Available=-
http://www.intuitivevoice.com/ pbxsales@intuitivevoice.com Phone 480-422-1244 Over ten years of telephony experience with fortune 100 companies. Specialize in turnkey Small Business PBX solutions. Focus on ease of use and tightly integrated solutions. We have a full support offering and full Asterisk management suite of tools. Many happy customers and references available. Visit our site for a complete product offering.
http://www.jaredsmith.net/asterisk-consultant.php Jared Smith asterisk@jaredsmith.net Phone 801-231-7154 Respected industry expert Co-Author of Asterisk: The Future of Telephony published by O'Reilly Media, September 2005 Experienced in Call Center design and installation Experienced in Custom Asterisk Programming and Troubleshooting Available for lectures and training Based in Utah, United States Fluent in Spanish; Hablo español!
Belgrade Serbia based information technology firm providing computer networking, Asterisk PBX support, Call Center and IVR solutions and consulting.

iPhonica provides consultancy for Next Generation VoIP Solutions. Provides development, maintenance, support, consultancy for Asterisk, SER. Complete end to end solutions like call centers, IP-PBXs, calling card application for service providers, session border controllers can be provided.

GEDAM Europe The European Arm of GEDAM Advanced Communications in New Zealand and Partner of SineApps (The producer of the Daily Asterisk News has finally gone public. The main base of operations for Europe is in Torino in the north of Italy, although we provide worldwide support. We specialise in all types of Asterisk and SER projects. We also develop embedded hardware platforms for running voice applications. We also have extensive experience with video over IP and can custom develop solutions to meet your needs perfectly.
         o Home page:: http://www.gedameurope.com
         o Telephone: PSTN +39 011 5694900
         o Email: info@gedameurope.com  We provide you with Asterisk-installations, VoIP-Consulting and customized programming.

Authorized Digium reseller for Russian Federation. Great experience with IP-PBX remote deployment and maintenance. Great experience with various IP-phones and VoIP-gateways. Asterisk security solutions, including Ranch Networks MIDCOM firewalls. DUNDi networks.

Divisão iComm
   * Phone: +55 (11) 4063-4304
   * Corporate IP Telecom solutions
   * IP Telecom consulting and Distributed communications
   * Asterisk consulting and advanced implementations.
   * Pre-Paid VoIP
   * Contact :
   * web: http://www.itfuture.com.br/
   * e-mail: icomm@itfuture.com.br  Mathias Kettner Asterisk and Linux Consulting and Training.
We offer a wide range of services and support concerning Asterisk, VoIP, XEN, Nagios, RPM, Softwaredistribution, Automatic Installation, AutoYast, Kickstart etc.

Professional VoIP and Asterisk consultant since 2005
Professional Linux SysAdmin, coder and security manager since 2000


Free & Open Source ICT solutions

At monoploid, we provide a plethora of ICT services based on free software and open technologies. We perform:
Asterisk installation Callshop deployment Corporate VoIP setups Scalable load-balanced/high-availability VoIP setups
Other VoIP services include:
Custom Asterisk development VoIP peering Remote technical support
Linksoft Technologies Ltd is a telecom software solution provider specializing in VOIP and traditional telephony applications. We provides consultancy, software development, and system integration services. We specialize in:
turn-key Asterisk-based IP-PBX software solution with an initutive web-based configuration and management interface custom Asterisk modules (e.g. outbound dialer, enhanced voicemail, IVR, etc) custom software development (web- or Windows- based) design, configuration, and installation of Asterisk PBX system

Website: Moki Systems Located: St. George, UT Email: Sales Phone Number: (435)-674-3571 The Moki Systems Team has experience configuring, maintaining and installing Asterisk Phone Systems. Our rates are very competitive and we offer our services in English and Japanese. The Moki Systems Team is Polycom Certified allowing you access to the highest quality phone systems at a reasonable price. Nasoni & Friends Located in Montreal Email asterisk.qc at REMOVEME gmail dot com Italian Number +39-02-303123373
My Friends and I have decided to offer our knowledge that we had developed around asterisk to help others. We are able to offer our services in Italian, English and French. We have a very competitive rates. NDSoftware provides VoIP consulting services.


We are a R&D oriented company, we have strategic alliances with several hardware resellers, focus on reliability and security We provide Consulting, Planning, Deployment, support and training services for Asterisk based PBX and appliances. Installation of Asterisk PBX with PSTN, VoIP, GSM and FAX integration Installation of switches with traffic management and Wifi Networks with wireless IP phones Turnkey solutions & Integration of asterisk with existing network utilizing SIP and/or IAX2. Call center solutions. Custom AGI scripts, diaplan extensions, billing systems Asterisk / VoIP Network Support & Management. 12x5x365 On-Call Engineer support services.
Contact Person:
S.Khan. Rao
CEO
Cell: +92.344.5900.154
Email: info@nethawk.com.pk
Company Website: www.nethawk.com.pk
NextGenData is an IP services company providing businesses the ability to completely re-engineer IT infrastructure using the most cost effective and reliable technology available. We provide SIP voice trunks and feature servers for Asterisk users. We also provide consulting on the VoIP architectures for multiple platforms including Asterisk. They focus on technology solutions for Start-up and Rapidly Growing Call Centers. The company has done extensive testing, troubleshooting and installation of asterisk for real world production environment. Its integrated technology package is complemented by a dedicated technical support team that provides expert business process support services to boost the client's competitive advantage in the highly dynamic call center industry.
NextIX technical support and development team are Digium certified professionals.
NextIX also operates an offshore call center and medical transcription company in the Philippines.
NeuralHead Diseño de redes Voip (Voip Network design). Consultoria Asterisk (Asterisk consulting). Guias de resolución de problemas (Troubleshoting guides). Soporte remoto de Asterisk (Asterisk remote support). Desarrollos a medida para Asterisk (Ad-hoc Asterisk development). Integración de sistemas (System integration). Miembro del AAUG (Member of AAUG).
Website:http://www.neuralhead.com
E-Mail:info@neuralhead.com Web site: http://www.aleph-tec.com/ E-mail: oleg.titov at gmail.com FWD: 74758 fwdOUT: 52442 Languages: Spanish, Russian, English Asterisk PBX, VoIP, SIP, H.323, worldwide remote administration, Linux sysadm, consultancy and training courses.
Special pricing for personal projects, SOHO/SMB clients. Secure and easy payment with PayPal. Solutions Designed :
CALL Center : Predictive , Priview and Progressive Modes
Asterisk : IP PBX / Audio Conferencing / IVRS/Speech Recognition with all indian languages
A2Billing
OpenSer : Sip Express Router
Building API to integrate and interact with your Inhouse Softwares
Hosted pbx,Hosted Ivrs.
Asterisk Training,
Sugar CRM
Prepaid Calling Cards
Wholesale Billing
Call Shop
Custom IVR Development
Hosted IVR Solutions
IP IVR Solutions
Enterprise Voicemail Systems
Carrier Voicemail Systems
Automated Dialers
Economic IP Phones (SIP / H.323 ) especially designed for Asterisk and SIP based pbx
Authorised Distributor for Rhino Cards in India
Aserisk and Freeswtich Cards available @
" http://www.telephonycards.co.in
We do Asterisk

Portland, Seattle and the Bay Area. Asterisk@Home solutions for small to mid-size business.
Asterisk based VoIP PBX and Call Center solutions with CRM integration; Video Phone, Video Conferencing and Video over IP to check, control and discuss face to face with your employees; Our consulting services will help your organization to covert its existing telephone system into cost saving, effective and flexible VoIP structure; On site and remote services from our consultants available everyday 8:30 - 20.00; We speak polish and english;

Specialising in the design, build and deployment of large scale telephony solutions. Progressive Voice Services provide several telephony solutions including operating one of the UKs leading conference call services.

sales@raynettech.com
Toll Free 877-693-2226
Local 860-693-2226
Full Implementation and Development of Asterisk and Cisco Phone Systems Custom Development Managed Cost Maintenance Contracts Full Service IT Support and Sales Call Centers VoIP over VSAT Emergency Communications Pre-packaged and Custom Systems Available
(Rodney Tamblyn) Dunedin, Otago, New Zealand Tel: (03) 4778606 x 1 Fax: (03) 4778606 Email: sales@sineapps.com Website: www.SineApps.com
US (NY, NJ): +1 800 824 6006, +1 212 537 6898
DE (Berlin): +49 30 868 70 9130
UK: +44 207 127 6310, +44 1510 601 2050
info@sipxx.com
sip:10888@fwd.pulver.com
Voice-over-IP, Asterisk PBX, SER, SIP Exchange, Snom4s, FreeSwitch SIP, IAX, MGCP, H323, SCCP, PSTN, IPv6 AudioCodes, Snom, Mediatrix, Sipura Design, Software, Integration, Installation, Support Custom SIP application development
Rafael A. Lama
Santo Domingo, Dominican Republic
Tel: (809) 620-2488
Email: info@smbconsulting.com.do
Website: www.smbconsulting.com.do Asterisk, Trixbox, Vicidial installation dCAP, FTOCC Certified
Melbourne, Victoria
Telephone +61 3 8888 5361 or +61 408 873 601
Email sales@stuartelvish.com
Specialised Asterisk systems for infrastructure projects including airports, roads, tunnels and carparks. Ad-hoc support or 24x7 remote support service level agreements available. Support for web, database and Asterisk integration.

Asterisk PBX installation, configuration and support. We also able to provide different solutions with advanced features. We offer remote support.
Melbourne, Australia. Integrated Open-Source Systems; since 1999. Installation and support of Asterisk and Linux server systems. Remote Support. Tel +61 3 9484 0013 email Robert Slater
IT and VoIP. Done right. Done now. (TM)
Worldwide remote consulting and support for Asterisk
General Asterisk/VoIP consulting Design, installation and maintenance of Asterisk-based PBX systems Call center infrastructure Custom application/reporting development Guaranteed satisfaction; many excellent references available Prompt service and response times Broad/in-depth team of telecom and data engineers
Contact: Dale Laushman
(303) 757-4611, Ext. 401
dale@theuptimegroup.com
www.theuptimegroup.com Sydney, NSW, Australia
Tel: +61 (2) 8090 0800

Product:
Mach 1 Asterisk-based PABX PABX Call Centre feature upgrades VoIP infrastructure suppliers
Consultancy:
CRM and CTI Integration services Design and Implementation consultancy VoIP Readiness Assesments
Managed Services:
Ongoing remote management and administration
http://www.ulterius.net/
sales@ulterius.net
Become your own Vonage with the VoIP Vendor solution. Everything you need in one asterisk box including a web site with customer interaface to view call history, billing, voicemail and custom settings. We are also a provider of te lowest rates on termination (outbound) and origination (inbound) for US and international calling. VoIP Reach also provides technical partnerships and professional project mamagement on VoIP Networks. Voiptech is an engineering company specialised in Enterprise Grade VoIP solutions for Providers, Telcos and Call Centers. Our portfolio included thousand of IP-PBX and Call Center installations. Voiptech is the developer of EasyPbx http://voip-info.org/wiki/view/EasyPbx an Hybrid Carrier-grade Software Platform to configure, control and monitor IP-PBX deployments in real-timebased on Asterisk and CallWeaver. Visit our web site, where you can find technical hints and further informations. Technical Contact : Giorgio Santini mailto:%20gsantini@voiptech.it Email : mailto:info@voiptech.it Telephone : +39 06 62 27 79 60 Home Page : http://www.easypbx.it
VoIP Consulting, Installation, Configuration and Programing for Aterisk, Yate, Web-Services and Apples MacOSX

Warrior Hill specializes in custom Asterisk solutions. Our team of developers can provide custom solutions for your Asterisk needs quickly and affordably. We provide a full range of remote Asterisk setup and management services.
Voices that bring asterisk to life
London, United Kingdom Tel: +44 (0)800 066 4864 (free phone) Email: sales(at)westany.com Website: www.westany.com
Westany provides international voice prompts for Asterisk. We currently have British English and French with German, Spanish and italian in production. All of the prompts can be extended by the same voice artists. Our pre-recorded packs start from just £149.99. Milano, Italy Tel: +39 02 40039010 Fax: +39 02 4151 203 Email: info(at)soluzionivocali.it Website: www.soluzionivocali.it
Provides you a wide range of Asterisk IVR solution for help desks, call centers, interactive voice applications and PBX integration.We can easily
interact and manage with database,TTS through the most modern tecnologies such SOAP,Corba,SIP,VoiceXML Located In Mumbai, India. Have implemented may Asterisk solutions for small and medium call centers and calling card shops. We have setup loadbalancing Vicidial Servers with high availability Mysql Cluster Database for high performance , stability and scalability.
 Phone +91-22-27581200
 Email dkumar at netsolutioninc.com
 Web : http://www.netsolutioninc.com
We provide a top tech services. Our main focus is in telecomunications and banking industry. Asterisk based solutions (IVR, Call center, billing, ...). Enterprise Application Integration via modern technologies. TintaDigital STI is a Portuguese company offering Information Technology services directed to all companies with technological needs.
Our team has solid skills in Asterisk systems, allowing the implementation of a set of services which goal is to support the planning and maintenance of the system's growth in a structured and consolidated manner.


We have extensive experience designing, installing and maintaining a different range of VOIP solutions. We can cluster and streamline your carrier solutions. We can help you design and implement you next big project. VoIP Experts’ team consists of qualified professionals with an extensive experience in VoIP area. The main spheres of our expertise are technical support, configuration, troubleshooting and system maintenancce of different VoIP solutions. We strive to provide highly qualified service along with fully satisfaction of our customers allowing them to achieve their key business strategies.
Phone: +15304051102 E-mail: sales@voip-experts.com
We offer various services for Asterisk such as SBC Configuration, custom live cd, wengophone customization, etc.
Contact us.

Dallas, Texas Sales@yellowboxsolutions.com Phone: 972-235-5800 Asterisk Consulting: Offer two different solution packages for SOHO and enterprise at economical prices. Our Consulting services also includes Customized Asterisk Development Asterisk configuration and installation Dialplan Designing AGI scripting IVR Designing Troubleshoot and other consulting services VOIP Hardware distribution include polycom and grandstream. see website for more details
Install and customize your VOIP PBX Solution.
Call Center Vicidial installation and integration.
Networking Linux Specialist

Radu Padure

www.voipexperts.com.ar / info@voipexperts.com.ar Location: Córdoba - Argentine Tel/Fax: +543515713406 Location: Buenos Aires - Argentine Tel/Fax: +541162612011 VoIP Consulting / Asterisk Specialists Design, Configuration,Integration, Installation and maintenance of Asterisk PBX Systems. We can provide installation and support of these systems worldwide. Custom asterisk PBXes for small, medium and enterprise level offices Custom business applications solutions and development based on * (Home Banking, Predictive Dialers, IVRs, CTI) Specializing in all facets of VoIP, Including provisioning, wholesale termination, high-volume Unified Messaging (Including Voice Mail, Fax Services, call conferencing, etc) Troubleshooting. Design. Analysis. Consulting
Guaranteed Results
Asterisk Consulting, integration, optimization and repair 24/7/365 Availablity Contact Center specialists with a proven track record Phone 602-505-4125 http://www.voipcitadel.com/ Creators of VicidialNOW distribution a call center suite automated installation pack in 1 CD!

Linux Server Installation (Web, Mail, etc...).
Asterisk Server Installation.

José Héctor Galimberti

Install and customize Asterisk PBX.
Using a powerful java swing GUI interface for User Management, CDR List, Call Recording, Call Spy, Server Health Monitor, Builtin IAX admin phone, Group Management, Remote Backup and Restore, Export every thing to excel. Call Center software and database integration. SMS & IVR Value added services provider.
Almontaha ICT Co.