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maple4VOIP

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Asterisk consultants Canada - Quebec

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514 DID's www.514dids.com (Montréal)


6447775 Canada Inc (Gatineau, Ottawa)


Aheeva Technology inc.

  • Web site: http://www.aheeva.com/
  • E-mail:info@aheeva.com
  • Phone: (514) 223-2581
  • Products: Inbound, SoftPhone, Predictive dialer, Full digital recording

Andre Courchesne - Consultant

  • Web site: http://www.net-forces.com/
  • E-mail:courchea@net-forces.com
  • Phone: (514) 241-2588
  • Products: Asterisk consulting, installation, custom programming, CTI integration, Broadcast dialer, Call Center dialer.

Atelka Contact Center Solutions

  • Web site: http://www.atelka.com/
  • E-mail:info@atelka.com
  • Phone: (514) 448-4905
  • Products: Virtual Contact Center

Asterisk Experts, Les experts en Asterisk

Montreal, Quebec, Canada

IVR

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What Is IVR?


IVR, or interactive voice response, is a what allows phone systems to process touch tones or voice waves during a telephone call. IVR technology is responsible for the menus people hear and respond to when they call up a company or business and hear the words: "press 1 for sales, press 2 for marketing, press 0 to speak to the operator," for example. IVR systems can be fully customized to play back dynamic audio, or pre-recorded menu options.

IVR is not necessarily related to VOIP, however, a VOIP IVR is. Most VOIP IVR systems or software support SIP based VOIP, but Skype IVR also support non-standard based Skype service.

Computer Telephony Component

IVR is an automated computer telephony integration CTI system which allows providers to create complex menus which the caller can navigate by using touch-tone key-presses or via spoken commands. IVR systems can be used as a Voice portal to access remote information such as bus scheduling where the caller can select the route for which they require information, or for billing or customer service systems which allow the caller to enter information such as their account number or credit card details without the need for operator assistance.

IVR and ACD Integration

IVR solutions are often integrated with an ACD, which routes incoming phone calls to agent work groups. This integration can be both a front end and back operation.

  • Most typically, an ACD system can route callers to an IVR program based upon DNIS or other parameters such as time of day or day of the week.
  • A smart IVR can transfer callers back to an ACD system to route the call to the next available agent within an agent hunt group.

One important task of an integrated IVR and ACD is to display Screen Pop information from the caller on the agent's workstation so that the agent has caller information readily available without the need to prompt the caller again.

IVR and Voice Broadcasting

IVR applications are typically associated with inbound calling programs. However, IVR technology can be applied to outbound calling campaigns and are most commonly used with Voice Broadcasting and touchtone responses. Examples of the application of this technology include the option to speak with an operator, opt out of a calling campaign, or taking an outbound survey.

Here is an example of IVR implementation in Voice broadcasting

Graphical Design Tool for IVR Applications

Recent IVR systems usually use high level scripting languages such as VoiceXML, an open standard for interactive voice response systems. For most users who lack technical training, developing an IVR system using scripting language, even high level language, are not feasible. The good news is there are design tools that are based on graphical user interface for the techies and none-techies alike. By using a GUI tool, a user can simply drag-and-drop components and create and deploy an IVR system in minutes. The whole design is a call flow diagram, much like a voicemail system user manual.

See Also (Vendor Information)

IVR Information


  • CCXML standard markup language for IVR / call control applications
  • IVR System Simulation Model - estimates resources required for an inbound calling campaign.
  • IVRS World - Blog about IVR

VOIP Event Calendar

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2015 VOIP Related Event:

May 2015



April 2015


March 2015

February 2015

  • 23 OpenSIPS eBootcamp - OpenSIPS online learning program 7 weeks of online classes and labs about SIP and OpenSIPS.
  • 11 -12 SACM 2015 - South Asian Carriers Meet (SACM) is the region’s leading corporate event, attracting the full spectrum of the telecom industry.
  • 9 -10 GCCM London 2015 - Carrier Community is organizing its Annual London 2015 GCCM taken place on 9th & 10th February 2015. Meet 500+ Club Members representing decision-makers from the Tier-1, Tier-2 and Tier-3 from 250+ operators in 40+ countries in London.

January 2015

  • 27 -30 ITEXPO Miami, Florida - The Business Technology Event, IT Expo East 2015, will be held in Miami, United States Of America on 27-30 Jan 2015 in Miami Beach Convention Center.

2014 VOIP related events.

For past years events see: VOIP Event Calendar Archive

December 2014

November 2014

  • 13 - 14 -

Asterisk High Availability Solutions

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This page outlines the various option available to create high availability for a VoIP PBX. Some are generic solutions while others are PBX specific. Some are complete HA solutions while others are half-baked scripts that do some things but not others.

Before you select a HA solution, carefully read this page on creating / selecting a High Availability solution (see Asterisk High Availability Design )


PBX Specific Solutions (Application Level Cluster)

These solutions provide clusters that are PBX specific. As noted on the Asterisk High Availability Design wiki page, these solutions create clusters at the Application level and are deeply PBX aware, environmentally aware, trunk aware, etc. The benefit of these solutions is that they are easy to install and provide complete clustering (heartbeat, data synchronizations, failure detection, sharing IP, etc.). The downside of these solutions is that they are PBX specific, so if your PBX software (eg: Asterisk, 3CX, FreeSwitch) is not listed below then you can't use these solutions.

  • HAAst (High Availability for Asterisk) from Generation D Systems adds high availability / clustering to any pair of Asterisk servers. The High Availability for Asterisk (HAAst) add-on offers rapid automatic failover of a failed peer, IP sharing, advanced peer health detection, intelligent synchronization of files and databases, etc. HAAst also supports manual promote/demote for maintenance, a command line interface, a telnet interface, a web based interface, and a developer API. Installation is straight forward, with no additional hardware required, no additional or complex heartbeat/cluster/etc software required either. HAAst is available in Free and Commercial editions, in use at call centers, hospitals, and other high-uptime environments. See High Availability Asterisk (HAAst) for more information.


Bundled HA Solutions (with OS Level Cluster)

These solutions use generic heartbeat/clustering at the OS level, bundled with a custom version of Asterisk. They use their own variant of Asterisk, custom GUI, etc. As noted on the Asterisk High Availability Design wiki page, these solutions create clusters at the OS level and are not deeply Asterisk aware, environmentally aware, trunk aware, etc. The benefit of these solutions is that they use off-the-shelf pieces to creat a cluster, the down side is they mirror corrupt data from one peer to the other, they don't detect deep Asterisk related & environmental failures, and they can be complex to administer (the HA). Some of these are clearly works in progress as well.

  • HiPBX is a new FOSS highly-available enterprise ready CentOS6/Asterisk/FreePBX distribution. It is currently being written by one of the original developers of FreePBX, and tightly integrates with (currently) Cisco SPA model phones. It provides Active Directory integration, automatic phone provisioning, user-configuration of phone features through their own Web interface, and cluster and node management through a GUI. This is all new software, and is not based off an existing project. It's currently under intensive development, and all source code is available on Github.

  • SARK-HA from Aelintra Telecom offers High Availability Asterisk out-of-the box. Runs Aelintra's SARK UCS MVP on a pair of servers.... Real-time failover takes less than 20 seconds to complete and includes support for ISDN PRI circuits. ...

Fail2Ban (with iptables) And Asterisk

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Fail2Ban


Fail2Ban is a standard Linux tool used to scan log files and then block IP's found in those log files using iptables. Fail2ban depends completely on the application (in this case Asterisk) to detect any intrusion/failure and log the user data, upon which fail2ban can then act. Fail2ban does not provide any type of intrusion detection, hack detection, etc., it depends completely on Asterisk to do that. As noted by Digium http://forums.asterisk.org/viewtopic.php?p=159984 fail2ban is not an intrusion detection / anti-hacking tool

Note that as of Asterisk 13 Digium is moving towards security events through the AMI, and moving away from log files. For now fail2ban is still compatible with Asterisk but consider fail2ban a short-term solution only. See this wiki page for alternatives: Asterisk security

You can get Fail2Ban, as well as more documentation, at www.fail2ban.org. At the time this is being written, the current release is 0.8.4.

Fail2Ban With Asterisk


The following describes how to setup Fail2Ban to work with Asterisk:

SECURITY NOTE: fail2ban is rather limited in its ability to detect attacks against asterisk.
More info http://forums.asterisk.org/viewtopic.php?p=159984
Consider a more comprehensive product like the free edition of SecAst www.generationd.com

Easy Install Script for Fail2ban version 0.8.4 / Red Hat


This script was written by Cédric Brohée in order to simplify and accelerate the integration of the solution in a basic Asterisk configuration on Red Hat.
Do not hesitate to read the bash script and make changes to match your own configuration.

Before running it, you will have to do chmod 755. ...

Asterisk security

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If you are looking to secure your PBX you have several options which can be implemented independently or in combination:
  • PBX Configuration - adjust the settings of your PBX to minimize obvious attack surfaces (no longer considered optional - just part of setting up any PBX).
  • Perimeter Security - Add software/hardware around your PBX to improve security (one notch above configuration - just part of operating any server).
  • Integrated Security - add software which integrates with your specific PBX to improve security (this is what really makes a difference in protecting your PBX).

Note that some recommendations (eg: changing ports, port knocking, etc.) are ideal for small and home office installations, whereas these same recommendations are impractical for large-scale implementations. As well, some recommendations are a great starting point (eg: hardware firewall) but this is no longer sufficient to protect a PBX.

Integrated Security


SecAst

SecAst is an intrusion detection and prevention system designed specifically to protect Asterisk phone systems against intrusion and fraud. SecAst uses a variety of techniques to detect intrusion attempts, halt ongoing attacks, and prevent future attacks. SecAst is available in three editions, including a free edition. SecAst can be downloaded from www.generationd.com or checkout the wiki page SecAst (Asterisk Intrusion Detection and Prevention)

Fail2Ban

Fail2Ban is a free utilitiy which looks at log files for records of failures (to register, etc.) and then add their source IP to iptables. See security warning regarding fail2ban http://forums.asterisk.org/viewtopic.php?p=159984 Fail2ban is not an intrusion detection / prevention tool, it depends completely on Asterisk to detect and reject an attempt from a hacker.


Perimeter Security

If you are looking to add layers around your PBX with generic protection:

Hardware Firewall

Most Asterisk boxes should be located behind a hardware firewall. Configure the firewall to block traffic from anyone that doesn't need to connect to you. Allow your VoIP provider, any remote phones/users, and others that may need to connect, but keep the restrictions as tight as possible. ...

Asterisk security through geographic IP address restriction

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Enhancing security through blocking of IP addresses on a geographic basis


There are many approaches to implementing security in Asterisk (see Asterisk security ), but for some system administrators it becomes a headache when certain users roam with a softphone or VoIP adapter and expect to be able to connect to the "home" Asterisk system from wherever they might be. Unless the user is connecting through a VPN, it may be impossible to know in advance what IP address they will be connecting from. This means you cannot preemptively set permit and deny settings, to only allow connections from an "approved" IP address.

While enforcing strong passwords can be very helpful in this situation, and the use of a separate Asterisk intrusion detection/prevention system is essential, additional security can be imposed through the use of geographic blocking. For example, if your users never travel outside their home country, then any connection from an IP address located outside the country would be considered extremely suspicious, even if the correct password is presented. Therefore, system administrators may want to consider automatically blocking connections from outside an "approved" area.

The purpose of this page is to list any scripts, software, or other mechanisms that attempt to enhance Asterisk security through the use of selective geographic blocking.

Available Software and Scripts


  • SecAst www.generationd.com is a product which can restrict Asterisk use based on Geographic IP location. It is compatible with IPv4 and IPv6, and allows you to restrict access by continent / country / region / city. SecAst is a commercial product but there is a free edition which is like fail2ban on steroids.
  • Geolock is a simple experimental Perl script that can be set up as a cron job to run once per minute. It does the equivalent of a "sip show peers" or "iax2 show peers" command from the Asterisk CLI, examines the IP address of each non-local connected extension, and uses a Perl module and geographic database to determine where that IP address is located. If the connection is coming from outside the home country (the US by default, but that is easily changed), then an IPtables rule is created that drops connections from that IP address. The extension itself is not banned, so the valid user should still be able to connect from within the "approved" geographic area.

  • Travelin’ Man is "a web- based, one-click Asterisk application that automatically reconfigures your Asterisk PBX to enable remote SIP phone access from your cellphone, iPad, remote PC, NetBook, or desktop telephone." It is said to only work with the "Incredible PBX" distribution. ...

TrixBox High Availability cluster using drbd

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Use this at your own risk. The script on this page is only intended to be run in a test environment, it will format partitions on your hard disk and is only intended to be run on a fresh install of Trixbox 2.6! For a more complete clustering solution check out Asterisk High Availability Solutions

I am not in anyway an expert in setting up clusters on linux, but thought I would share my little script in an attempt to get some feedback for improvements and to share with other users wanting to try setting up a similar system.

I am testing this on two identical Dell Poweredge R200 servers installing Trixbox using the advanced option so I could specify the disk partitions I wanted to create.

/dev/sda1 /boot
/dev/sda2 /
/dev/sda3 /share
/dev/sda4 swap

The server has two network interfaces
eth0 is used to communication to the LAN
eth1 is used for DRBD disk replication and is connected via a crossover cable


Enjoy! - Royce

References Used:
TrixBox Forum Post
Elastix Forum Post



#!/bin/bash

# Make sure you edit this section to your requirements! Start
cluster_ip='192.168.16.30'
domain_name="domain.local"
gateway='192.168.16.254'
primary_dns='192.168.16.2'
secondary_dns='192.168.16.2'
master_hostname='asterisk-master'
master_ip_address_eth0='192.168.16.31'
master_ip_address_eth1='10.10.10.1'
slave_hostname='asterisk-slave'
slave_ip_address_eth0='192.168.16.32'
slave_ip_address_eth1='10.10.10.2'
subnet_mask_eth0='255.255.255.0'
subnet_mask_eth1='255.255.255.0'
drbd_disk='/dev/sda3'
drbd_device='/dev/drbd0'
# Make sure you edit this section to your requirements! End

clear
echo "TrixBox High Availability Installation Script"
echo
echo "1. Master node"
echo "2. Master node (pause after each section)"
echo "3. Slave node"
echo "4. Slave node (pause after each section)"
echo "5. Exit"
echo
echo -n "Please select the installation type. "

keypress="0"
until [ $keypress = "1" ] || [ $keypress = "2" ] || [ $keypress = "3" ] || [ $keypress = "4" ] || [ $keypress = "5" ]; do
  read -s -n 1 keypress
done

case $keypress in
  1)
    node='master'
    debug=0
    server_hostname=$master_hostname
    server_ip_address_eth0=$master_ip_address_eth0
    server_ip_address_eth1=$master_ip_address_eth1
    ;;
  2) 
    node='master'
    debug=1
    server_hostname=$master_hostname
    server_ip_address_eth0=$master_ip_address_eth0
    server_ip_address_eth1=$master_ip_address_eth1
    ;;
  3)
    node='slave'
    debug=0
    server_hostname=$slave_hostname
    server_ip_address_eth0=$slave_ip_address_eth0
    server_ip_address_eth1=$slave_ip_address_eth1
    ;;
  4)
    node='slave'
    debug=1
    server_hostname=$slave_hostname
    server_ip_address_eth0=$slave_ip_address_eth0
    server_ip_address_eth1=$slave_ip_address_eth1
    ;;
  5)
    echo
    exit
    ;;
esac

clear
echo "Installation will continue with the following settings:"
echo
echo "Cluster IP address -" $cluster_ip
echo
echo "Node name -" $server_hostname.$domain_name
echo "Node IP address (eth0) -" $server_ip_address_eth0
echo "Node IP address (eth1) -" $server_ip_address_eth1
echo "Node Subnet mask (eth0) -" $subnet_mask_eth0
echo "Node Subnet mask (eth1) -" $subnet_mask_eth1
echo "Node Gateway -" $gateway
echo "Node Primary DNS -" $primary_dns
echo "Node Secondary DNS -" $secondary_dns
echo "Master node name -" $master_hostname. ...

Cisco

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Cisco Systems, Inc

Cisco's IP telephony products offer great flexibility, a large feature set, and stand up well to every day corporate use. That being said, they're also not the cheapest in the field.

Products


Most likely you'll need access to firmware packages for upgrades or version changes, i.e. converting a phone from SCCP to SIP. The files are available on Cisco's website along with a good amount of documentation but you'll need a service contract. The service contract for a Cisco phone is around $8/year. The service contract price will be based on the value of the item you want covered. Cisco has several support options. You just need the most basic support in order to be able to get updated software.



GKTMP

GKTMP is Cisco's open, but proprietary, protocol used for communication between a call control device such as a Cisco Gatekeeper or SIP Proxy and an external application such as a route server. GKTMP, which stands for GateKeeper Transaction Message Protocol, was originally developed for H.323 gatekeepers but since it is a rich and lightweight Operations and Billing Support System (OSS/BSS) protocol, it is equally useful as an API to external applications for SIP proxies or B2BUAs. A major benefit of GKTMP is the ability to offload complex routing algorithms, such as least cost routing tables with hundreds of thousands of routes, to an external server.

An open source module which provides a GKTMP interface to open source OSP servers, such as OpenOSP and RAMS on www.sipfoundry.org/OSP, is available at sourceforge.net/projects/gktmp-to-osp.

Service Contract


Obtaining a service contract directly from cisco can be a lot of fun. When last checked, there's no online registration available, only a page with an email address that's no longer valid. ...

Asterisk Paid Support

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Asterisk Consultants Romania

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Add your entry here (Alphabetical order by country and company):
This page is growing large. Please don't post logos!!


Artvister

  • Web site: http://www.artvister.com
  • Phone: +40 (0) 332 408 782
  • Fax: +40 332 819 586
  • E-mail: office@artvister.com
    • Asterisk PBX Installation and Support
    • PBX - VOIP integration and maintenance
    • Telco Support integration - ISDN PRI/BRI
    • Call Center, Contact Center Support Services,Data Entry.
    • Professional Technical Support Services
    • Auto Dialer, Predictive Dialer, ACD Services,Billing Services.
    • Mediant,AudioCode,Digium,Topex,2n integrator.
    • Microsoft Lync, Kamailio/Opensips
    • Software Development and Maintenance
    • Ticketing,Network Monitoring Services.
    • IVR,TTS.
    • VirtualPBX, Hosting PBX, CTI and CRM Integrator.
    • Cloud PBX, Cloud Vicidial, ViciBox, AgileDial
    • Phone Booking App.
  • Support Contracts

Dako SRL

  • Web site: http://www.dako.ro
  • E-mail: office@dako.ro
  • Phone: +40 722 274222
  • Fax: +40 356 815819
    • Asterisk installation, configuration & customization
    • Call center solutions
    • Development (IVR, AGI)

Fluid Code SRL


Inovo Solutions

  • Web site: http://www.centraleip.ro
  • Phone: +4 031 8282 200
  • Fax: +4 031 8282 209
  • Fax: +4 021 4107 088
  • E-mail: office@inovo.ro
  • Services:
    • Asterisk installation, configuration and suppport
    • Call-Center solutions
    • Development (IVR, AGI, AMI)
    • Call-center solutions
    • PBX trunking

Iptelis Networks

VOIP Consultants

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Grandstream

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Grandstream is a manufacturer of SIP VOIP products.

Products

SecAst (Asterisk Intrusion Detection and Prevention)

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SecAst-logo.png

Overview

SecAst (Security For Asterisk) is an intrusion detection and prevention system designed specifically to protect Asterisk based phone systems against attack and fraud. SecAst uses a variety of techniques to detect intrusion attempts, halt ongoing attacks, and prevent future attacks. In addition, SecAst uses advanced techniques to detect valid credentials that have been disclosed / compromised and are being abused. SecAst also uses heuristic algorithms to detect fraudulent activity based on known attack patterns. Upon detection SecAst blocks the current attacker from the Asterisk host at the network level. SecAst can also allow/deny any user based on the greographic source (country / region / city) of their IP address.

SecAst is a 100% software solution, communicating with Asterisk primarily through the Asterisk Management Interface (AMI), but also monitoring Asterisk message/security logs for relevant information, and also communicating with the Linux network interfaces. The data from these sources allows SecAst to monitor connection and dial attempts with invalid credentials, the rate at which users/peers are dialing, the number of channels in use by user/peer across all protocols, the source IP of remote users/peers, etc. By combining this data SecAst can effectively stop attacks/fraud in its tracks, and alert the administrator with details of each attack.

SecAst offers detailed geographic allow/deny rules (geofencing) down to the city level without large or complex firewall rules (all geofencing rules remain within SecAst). Use of geofencing dramatically reduces the number of, and risk from, attacks, allowing administrators to quickly eliminate continents/countries/regions/cities where their users would never be located.

SecAst offers extensive interfaces to interact with other programs, utilities, external firewalls, billing systems, etc. allowing for considerable customization. For example, changes in Threat Level can trigger scripts which alert administrators, shutdown interfaces, change firewall rules, etc.

SecAst is available in both free and commercial editions. You can get SecAst, as well as more documentation, at www.generationd.com.

technology_overview.png



Asterisk Compatibility

SecAst is compatible with a broad range of Asterisk versions and distributions. SecAst works with Asterisk versions 1.4 through 12, both 32-bit and 64-bit. SecAst is also compatible with a wide range of Asterisk distributions, from Digium's plain old Asterisk, to FreePBX and PBX In A Flash and TrixBox, to 3rd Lane and more. ...

High Availability Asterisk (HAAst)

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HAAst-logo-mini.png


Overview

HAAst is software package designed specifically to create a high-availability cluster out of an any pair of Asterisk servers. HAAst can detect a range of failures on one Asterisk server and automatically transfer control to the other server, resulting in a telephony environment with minimal down time.

HAAst is a 100% software solution, with switchover in seconds. Built-in intelligent network control allows for a single IP address to be shared between peers, so clients/phones automatically connect to the active Asterisk server without change. Built-in replication and synchronization between servers also reduces maintenance and support activities.

HAAst is an easy to use solution, with shell (command line), telnet, socket, and web interfaces, suitable for beginners and experts alike. HAAst is ideal for demanding telephony environments like call centers, emergency 911 operations, medical facilities, and mid-to-large size businesses, as well as small-businesses looking for high PBX uptime using low-cost off the shelf components.

HAAst does not require any other High Availability/heartbeat software, nor require use of shared disk/block level disk sharing etc. HAAst also does not require any other specialized hardware so there is no single point of failure. HAAst is robust in functionality but simple to set up and use.

HAAst is available in Free and Commercial editions. The Free edition is suitable for companies wanting to test if the basic functionality & compatibility meets their needs. In addition, the Free edition is a functional and useful high availability add-on for SOHO environments, more capable that any other DIY scripts, etc. The commercial edition is suitable for companies with critical telephony uptime requirements including call centers, 911 emergency centers, hospitals, etc.

technology_overview.png



Asterisk Compatibility

HAAst is compatible with a broad range of Asterisk versions and distributions. HAAst works with Asterisk versions 1.4 through 13, both 32-bit and 64-bit. HAAst is also compatible with a wide range of Asterisk distributions, including Digium's plain old Asterisk, FreePBX, PBX In A Flash, TrixBox, 3rd Lane, and more. HAAst can even control a custom distribution through settings which allow starting and stopping any executable. ...

Cloud Telephony Management System (CTMS)

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medium1.jpg


What is CTMS?

CTMS:abbreviation of Cloud Telephony Management System, is the perfect solution. CTMS consists of three parts: CTMC+ CTN+ IP Phone.

CTMC: (Cloud Telephony Management Center) is a network node management center which has been independently developed by ZYCOO, and can be utilized by VoIP service providers and enterprises users to manage multiple CooVox CTNs (Cloud Telephony Nodes). CTMC provides a multitude of features for CTN including auto-provision, software/firmware upgrade management, status/performance monitoring, warning log diagnosis etc. CMTC is a powerful solution that delivers the features and functionality required to manage and maintain a highly dispersed telephony environment through the use of a single centralized management system.

CTN: (Cloud Telephony Node) is the node of cloud telephony management system and handling the switch of telephony communication. ZYCOO CooVox series IP PBX can be taken as the node after upgrading. CTN is allowed for branch’s administrator to configure the local network connections.

IP Phone:which support SIP protocol can be used in CTMS; especially the phones support auto-provision with ZYCOO CooVox Series IP PBX; all the phones located in different CTN are allowed to be auto-provisioned via CTMC directly.

Benefits & Features:

  1. Centralized configuration and upgrading of CTNs
  2. Monitor system information, configuration and service status
  3. View and backup of system log, operation log and call log
  4. Manage multi-service and user groups based on template
  5. Manage configuration for individual or multiple devices based on user groups
  6. Flexible upgrading control strategy allowing for convenient software and firmware upgrades
  7. Based on TR069 protocol, allowing nodes to pass through private networks
  8. Adopting B/S managing mode to achieve multi-language GUI, humanized management process, and easy operation
  9. Based on Linux which ensures the device is secure and reliable
  10. Password change supported and license authentication available
ctms.jpg
medium3.jpg
medium4.jpg


Contact us:



ZYCOO China
Web: www.zycoo.com
Tel: +86 (28) 85337096
Address: 7F, B7, Tianfu Software Park, Chengdu, China.


ZYCOO UAE
Web: www.zycoo.ae
Tel: +971 (4) 3798839
Address: #305, Al Riffa Plaza Bldg., Khalid Bin Alwaleed RD, DUBAI, UAE


ZYCOO UK
LS21, Armstrong House, First Avenue, Robin Hood Airport, Doncaster, UK (DN93GA)



ALL THE RIGHTS ARE RESERVED BY ZYCOO Co.,Ltd

Mobile VoIP

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Mobile VoIP is an efficient, low-cost way to communicate using your cell phone and the services provided by your home or business VoIP provider.

How Does Mobile VoIP Work?


Mobile VoIP works with a cell phone’s 3G, 4G, GSM, or other Internet service to send voice calls as digital signals over the Internet using voice over IP technology. Mobile VoIP phones can also take advantage of WiFi hotspots to eliminate the calling costs of a cellular voice or data plan.

By using VoIP, mobile VoIP phone users — especially smartphone users — can benefit from lower costs when calling, texting, or other common smartphone activities. Digital data transmission using VoIP is also typically faster, as the data is spread out over multiple packets, each taking the fastest route to its intended destination.

Using a mobile VoIP phone with WiFi hotspot access can also reduce a mobile VoIP phone user's costs by sidestepping the carrier's expensive 3G service altogether. For instance, with a cellular carrier's monthly data plan, callers can easily exceed bandwidth maximums, incurring overage charges. Tapping into WiFi hotspots with mobile VoIP software reduces that risk and extends the lifespan of the monthly data allotment.

A mobile VoIP phone service can eliminate the need for a basic voice plan, as well as optional (and costly) text add-ons. With a mobile VoIP phone, cell phone users can enjoy more flexibility in calling times than a cellular voice plan provides, with fewer restrictions. VoIP mobile phone service means that a mobile VoIP user can make unlimited inexpensive or free calls using voice over IP technology at any time.

Mobile VoIP users don't need to worry about the limitations associated with cell phone calling plans, such as:

  • Anytime minutes
  • Night or weekend minutes
  • Rollover minutes
  • Roaming charges
  • Incoming call charges
  • Messaging limits
  • Mobile-to-mobile calling (check with your mobile VoIP provider, some do treat in-network calls differently)

Mobile VoIP phone users can also take advantage of the additional, integrated features a mobile VoIP app supports. This includes high-bandwidth activities such as group chat and video chat. Accessing these functions without mobile VoIP software (by fring or Talkonaut, for instance), typically requires a separate app, and using it could impact or exceed monthly text and bandwidth maximums.

Accessing Mobile VoIP

Cell phone users can use mobile VoIP service on their phone with the addition of mobile VoIP software. These are apps offered by VoIP phone service providers customers may already be using at home or at work, such as Vonage, or standalone mobile VoIP apps such as Skype, Vyke, or Truphone.

Some services, such as Truphone, also offer an entire mobile VoIP network by combining a SIM (Subscriber Identity Module) card and an app together. (The SIM card contains all the information needed to identify network subscribers. ...

ICTDialer

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ICTDialer Is an Open Source Unified Communications auto dialer Software

ICTDialer is an open source Unified Communications marketing Software. ICTDialer is multi-tenant with Voice, SMS & Fax broadcasting capabilities developed over re-known open source Content Management System Drupal and Freeswitch based powerful ICTCore Communication Framework . It can be scaled to blast thousands of simultaneous calls using either VoIP, Foip or PSTN. ICTDialer capable to fit in many broadcasting and telemarketing scenarios. It empowers user with capabilities of Drupal CMS and ICTCore Communication Framework

ICTDialer is released as Open Source GNU AGPLv3

ICTDialer is developed and promoted by ICT Innovations with in-depth experience in open source ICT's , Unified Communications and Telemarketing technologies.

Today, January 3, 2013 , Released new version of ICTDialer Version 1.0.0 with following additional features

  • Improved Trunk management support
  • Select Trunk in campaign form *
  • Select custom caller ID in campaign form
  • Set Max call duration in campaign form * Bug fix in campaign scheduling
  • Added Agent/Extenions management support *
  • Added support for IVR Call transfer Application
  • Real-time Campaign Report updates
  • Sample import contact CSV file at GUI

Hosted PBX

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Hosted PBX is a service where the call platform and PBX features are hosted at the service provider location. The business end users connect via IP to the provider for voice service.

"Hosted" means to say that the hardware and PBX is hosted at an off-site location from where the VoIP telephone service is being used. An office can have VoIP telephone service that powers their phones in the office, but their PBX could be hosted at their VoIP providers data center, thus the term: hosted PBX. Hosted PBX is also sometimes referred to as Hosted VoIP.

Benefits of Hosted PBX


There are many benefits to using Hosted PBX rather than a traditional phone system, or an on-premise PBX. The main benefit is cost- a Hosted PBX system costs much less to set-up than an on-premise PBX. In many cases, there are no set-up fees for a hosted PBX system. Purchasing and setting up an on-premise PBX can cost tens of thousands of dollars. Hosted PBX phone systems fall under operational expenditure rather than capital expenditure, which also makes hosted PBX service attractive to businesses. With hosted PBX service, you pay a monthly fee, and the hosted PBX service provider takes care of the rest.

Another benefit to a hosted PBX system over an on-premise PBX is that hosted PBX service providers will take care of all the set-up and installation, meaning you do not need to be a telecom or VoIP expert in order to get a hosted PBX system. A downside to a hosted PBX is you may have a little less of an ability to customize your solution to your business, but many hosted PBX service providers can achieve a deep level of customization.

More Hosted PBX Service Providers



OnePipe

1Pipe Telecom | OnePipe is a licensed carrier providing Hosted PBX and SIP Trunking in over 50 countries. We have regional network facilities spread across the globe.
  • Actual US CLEC
  • Branded customer portals
  • Multiple geographic locations on one Hosted PBX
  • Coverage in over 50 countries
  • Unlimited inbound on each channel
  • Great for inbound call centers
  • No fee's per user or extension



Arrrowtel - Increase efficiency with a Hosted VoIP solution. Our rates are competitive, our attention to detail borders on the obsessive. No-charge, on-site installation.


CebodTelecm - Cebod Telecom offers Cloud based hosted Business Telephone system for small to mid size businesses. No need to invest expensive hardware and software, fully managed and scalable system take away your phone system worries. To learn more about VoIP check us out at http://cebodtelecom.com/

Most common features Included in your plan
  • Auto Attendant
  • Find Me/Follow Me
  • Ring Group
  • Call Queues/Call Center
  • Time Based Routing
  • Conferencing
  • Voicemail
  • Voicemail to E-mail or Fax
  • High Definition Calling
  • 800/Toll Free Numbers
  • Local numbers across the US, UK and Canada
  • Unlimited Calling
  • Unlimited Extensions
  • Unlimited Local/Long Distance
  • Call Logs
  • Call Recording
  • Number Portability
  • No Setup or Activation Fees
  • and SO much more

Advanced phone solutions, tailored for your business. Cebod Telecom is a professional, reliable phone service provider that improves your company’s productivity.



Svanto.net - Tailored Internet telephony solutions
VoIP provider for residential, wholesales and business. ...
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