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Toll Free Termination Providers

March 12, 2015, 2:24 pm
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Toll-free termination providers allow you to terminate toll-free calls from the US and Canada for free. If you have a large volume of calls to toll-free numbers, some providers will pay you for your calls. Carriers who have direct agreements have a higher success in collection from LD carriers who support the RespOrgs CIC.

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Without registration required


HyperCube Leading provider of Wholesale Toll free termination with Collection allowance.
  • Free SIP termination to all NA 8YY destinations.
  • Free SIP trunk from any GLOBAL IP.
  • Free Reporting and Stats on all traffic.
  • Contract required for Collection allowance.
  • CLECs are welcome, MECABS available.
  • Tandem Replacement and full compliance.
  • No Dialer Traffic Allowed.
  • Free Toll Free Numbers for Toll free services available as well.

TollFreeProxy.com - Free Toll-Free SIP Termination
  • Free SIP termination to NANPA toll-free destinations ( 1-800, 1-844, 1-855, 1-866, 1-877, 1-888 )
  • Caller-ID / ANI is delivered as sent
  • Crystal clear audio with minimal latency and jitter
  • G711u / ULAW with 10ms packetization
  • SIP Registration NOT required!
  • Customer Support Available via E-Mail
  • Compatible with postage meters!
  • Compatible with 10 digit, 11 digit, and full E.164 number formats

Alcazar Networks Inc - Free toll-free termination
Alcazar Networks - VoIP Services
  • Providing FREE toll-free termination to the US48 800 - 855 - 866 - 877 - 888
  • Full accurate Caller-ID number (ANI) delivered
  • Codecs supported: G711 and G729
  • NO SIP REGISTRATION REQUIRED
  • If you require registration we accept ANY credentials so you can start passing calls immediately.

ArcTele Communications, Inc - Toll Free Termination
http://www.arctele.com
ArcTele Communications, Inc offers a Toll Free termaintion gateway service free of charge.
- We send your callerID
- ULAW and G729
- Send calls in the format of 18XXXXXXXXX
- Unauthenticated to TOLL FREE NUMBER@tf.arctele.com port 5060

Broadvox is a leading wholesale VoIP service provider that delivers reliable VoIP solutions for domestic and international businesses.

Broadvox has 8YY number toll-free termination services that provide switching for 8YY calls originated by customers worldwide; conduct the database lookup and determine to which Interexchange carrier (IXC) the toll free number belongs; then route the call appropriately for call completion.

For more information, to
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Session Border Controller

March 13, 2015, 9:02 am
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A Session Border Controller is a device used in select VoIP networks to exert control over the signaling and usually also the media streams involved in setting up, conducting, and tearing down calls. The SBC enforces security, quality of service and admission control mechanism over the VoIP sessions.

The Session Border Controller is often installed in a point of demarcation between one part of a network and another. Most Session Border controllers will be installed between peering service provider networks, between the enterprise network and the service provider network, or between the service provider network and residential users.

A Session Border Controller is like a Firewall for VOIP.
They are often configured as a SIP Back-To-Back User Agent (See SIP RFC).

In addition to firewall functions they also may provide services like NAT traversal.

See: NAT and VOIP for more information and watch the video on The Anatomy of Session Border Controllers and To Couple or Decouple Routing Intelligence from SBC.

Martyn Davies of Dialogic discusses Session Border Controllers as the often misunderstood "black magic" application for VoIP networks.

Although carriers often use Session Border Controllers for signal translation and security, most do not include the hardware-based signal processing needed for media transcoding. For all-IP environments, new elements are required that can mediate signaling, transcode among different media formats, and handle basic security issues. The concept of Multimedia Border Element (MMBE) meets these needs.

Products


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New Software Releases

March 16, 2015, 5:33 am
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This page is to inform on various VoIP related software releases.

Your contributions are welcome, but please read the How to add information to this wiki page and the Posting Guidelinesbefore you post.

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March 2015


February 2015

  • 2015-02-25 -
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Asterisk consultants USA

March 16, 2015, 4:42 pm
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This is a comprehensive list of Asterisk consultants in the USA (United States). Add your entry here (alphabetical order, by state and company), but stick to states where you have actual presence!

Feel free to add a few lines (max 5) describing your business. Don't forget to add VoIP telephone numbers, like a SIP URI. Use common courtesy with others' entries! No images!

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ALABAMA


Asteria Solutions Group

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VOIP Consultants

March 16, 2015, 7:33 pm
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Asterisk system vendors INDIA

March 16, 2015, 7:48 pm
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This is a list of Asterisk system vendors in India.

AAB Asterisk Consultant India


Services Provide:

  • Predictive Dialer
  • Voice Broadcasting & Fax Blasting
  • 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
  • vtiger vicidial integration

IP PBX/ Installation / maintenance / configuration of linux systems / servers VOIP Gatekeepers / Phones / devices.

Support for digium / openvox / sangoma / rhino E1 / PRI / FXO Digital / analog Telephony Cards /grandstream

Asterisk Hyderabad Ph : 9392335385
Asterisk Bangalore Ph : 9392335385

Asterisk IPPBX India (Hyderabad)

Phone: +91 9392335385

Asterisk Consultant : Subramanyam (subbu)

subbu6699@gmail.com

Grandstream hyderabad Ph : 9392335385
Asterisk Chennai Ph : 9392335385
Asterisk Mumbai Ph : 9392335385
Grandstream india Ph : 9392335385
predictive dialer hyderabad Ph : 9392335385
voip hyderabad Ph : 9392335385
Asterisk Service and Solution

  • Asterisk Services and Solution Provider division of Ecosmob Technologies Pvt. Ltd. has expertise in open source and Asterisk PBX. The company has developed many custom software solutions for the global clients and enjoy the leading position in the Asterisk industry. The team of experienced Asterisk developers offers design, development, configuration and support services for following
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WebRTC

March 16, 2015, 8:10 pm
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Synopsis

The practical implementation of VoIP was started on hardware based IP Phones. The idea was well received and was transferred into the concept of Soft Phones or software based IP Phones. These softwares always required some additional installation to the native Operating System. Most common examples of Softphones or Software based SIP client is Counterpath's X-Lite and Bria.

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back to top

The Evolution of Software Development made it possible to translate or formulate equivalent of almost every desktop based application to web based application. This brought major shift in Software Industry as the web browsers are integral part of almost every Operating System. SIP clients, were also transformed into Web Extensions. Most of the time, Flash was used to develop such extensions however, it always required extra plugin installation, thus decreasing system performance, and increasing chance to troubleshoot as it required additional resources to be deployed. And this problem gave rise to the concept of WebRTC.

Overview

customLogo.gif.png

WebRTC provides the functionality of realtime multimedia applications without any installation of additional plugins, downloads or extensions. The ideal form of WebRTC describes such web based Real Time Communication independent of Browser being used by user. It's a Javascript based API originally being developed to develop browser to browser communication applications for Voice, Video and Peer to Peer File Sharing tasks.

Architecture

The architecture of WebRTC, as described by W3C looks something like this:
WebRTCpublicdiagramforwebsite (2).png


Design

Major components of WebRTC include:

  • getUserMedia, which allows a web browser to access the camera and microphone
  • PeerConnection, which sets up the audio/video calls
  • DataChannels, which allow browsers to share data via peer-to-peer

Support

Chrome WebRTC Development Team

Discussion List: https://groups.google.com/group/discuss-webrtc
Google Plus Page: https://plus.google.com/113817074606039822053
Chrome WebRTC Issue Tracker: http://code.google. ...
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Unified Communications

March 16, 2015, 8:20 pm
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Unified Communications, often abbreviated as UC, is the combination of different types of communication mediums including telephone service (VoIP), chat, video and web conferencing, messaging, email, fax, and other types of communications. Unified Communications systems can be sold and implemented with a select amount of individual communication mediums, or many integrated into a single unified system.

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Unified communications consist of integrating various real-time and non-real time communication services into a convenient package for the user. Examples of real-time services commonly used include instant messaging, telephony and data sharing. Non-real time communication services focus mainly on messaging services like e-mail, SMS, voicemail and fax.

With a unified communications solution, the user can receive a message through one communication medium and use another one to access it. One common example of this would be receiving a voicemail message on an office landline phone and using a mobile phone to retrieve it. ...
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voip-info.org

March 17, 2015, 2:19 am
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Welcome to the VOIP Wiki - a reference guide to all things VOIP.


This Wiki covers everything related to VOIP, software, hardware, VoIP service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony.

Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelinesbefore you post.


NEWS


News Resources


Getting Started

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Cloud Telephony Management System (CTMS)

March 18, 2015, 12:43 am
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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
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IP Office for SOHO

March 18, 2015, 12:45 am
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UC510-1.jpg
UC520-1.jpg


IP Office for SOHO


UC510/UC520 is designed as IP Office for SOHOs (Small Office and Home Offices)specially. The new solution offers not only a Wi-Fi router supporting 3G/LTE data access (UC520), ADSL, VPN Client/Server, VLAN... but also a fully featured IP PBX that can host up to 10 extensions with 2 analog ports connected (PSTN line (FXO) / analog phone (FXS)), and supports Call Forward, Call Recording, Blind/Attendant Transfer, Conference, and so on.

UC510/UC520 is configured and managed through a single web GUI which significantly reduces the time and effort required to install the product.This simplified management and reduction in hardware costs through merging two products into one makes the UC510/UC520 an amazing and cost effective solution for SOHOs.






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
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CooVox Series IP Phone System

March 18, 2015, 12:45 am
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U-1small.jpg


Highlights:


No License Fee
Advanced IP PBX features
Automatic Call Distribution(ACD)
Call Detail Records(CDR)
Fax to Email, Email to Fax
Voicemail to Email
Industry SIP Trunk and Digital Trunks Supported
Firewall Intrusion Detection
HD Voice(G.722.) Supported

Applicable Trunks: FXO/ FXS/ GSM/ BRI/ E1/T1

Software: Developed based on OpenSource Asterisk 1.8.x

Hardware: Dualcore CPU; Modular Design (flexbile to change the module)

CooVox Series IP PBXs come in four sizes: U20 / U50 / U60 / U100

CooVox-U20: Support 30 Ext. Users, Max. 10 Concurrent Calls
SDRAM 128MB DDR2; Memory 4GB SD Card

CooVox-U50: Support 100 Ext. Users, Max. 15 Concurrent Calls
SDRAM 256MB DDR2; Memory 4GB SD Card

CooVox-U60: Support 200 Ext. Users, Max. 80 Concurrent Calls
SDRAM 1GB DDR3; Memory 32GB SSD

CooVox-U100: Support 500 Ext. Users, Max. 80 Concurrent Calls
SDRAM 2GB DDR3; Memory 500GB HD


U100-5.jpg

U-6.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
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Asterisk system vendors

March 18, 2015, 12:54 am
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Asterisk timer ztdummy

March 18, 2015, 4:06 am
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How to compile ztdummy

ztdummy is a Linux kernel module that will provide your Asterisk with a Zaptel timer even if you don't have any Digium hardware installed in your Linux server. For Linux kernel version 2.4 the ztdummy module takes timing from the usb-uhci kernel module, which should be loaded before ztdummy. Note that usb-uhci must not be compiled into the kernel, it'll only work with ztdummy if loaded as a module! On kernel version 2.6 ztdummy uses internal high-resolution kernel timer and does not require any USB. Using the internal kernel timer is recommended.

General instructions for compiling all of the zaptel modules including ztdummy are here: Asterisk Zaptel Installation

Do you have the correct USB Controller?

If you are using a Linux 2.4 kernel you will need a USB controller to use ztdummy.
There are two types of USB controller chips used on motherboards: USB OHCI and USB UHCI. To use ztdummy, you need an USB UHCI type controller on the motherboard, as the OHCI chips work very differently. If you do not have an appropriate USB controller as a timing source, then you should use Asterisk zaprtc. It was originally written by Klaus-Peter Junghanns and is distributed at http://www.junghanns.net/asterisk/.

  1. To check if you have the usb_uhci module, do lsmod
  2. Check out the zaptel module from the Asterisk CVS repository
  3. Make sure you have the Linux Kernel source files installed
  4. Edit the Makefile and remove the '#' in front of ztdummy in the top
  5. Do a make all
  6. Do make install
  7. To load the ztdummy, do modprobe ztdummy

You will probably want to include 'modprobe ztdummy' in your /etc/rc.d/rc.local to make sure it is present at startup before Asterisk is launched.

Additional info


This page contains outdated information. Now that 2.6 kernels are norm, it still talks about 2.4 and 'make linux26' which have been deprecated.
Also these instructions do not work on Fedora 6/7 as of kernel 2.6.22.1-32 . These are the errors when doing modprobe ztdummy after compilation
ztdummy: Unknown symbol rtc_register
ztdummy: Unknown symbol rtc_unregister
ztdummy: Unknown symbol rtc_control

even though crc_ccitt is set as M. here is the output from lsmod | grep zaptel
zaptel 186404 0
crc_ccitt 6337 1 zaptel

There is a bug opened with Digium

http://bugs.digium.com/view.php?id=10426

Install instructions for Debian Lenny as of 2/3/2010 for ztdummy timer

Check out zaptel package
svn checkout http://svn.digium.com/svn/zaptel/branches/1.4 zaptel
Install kernel source (assuming you've got build-essential)
sudo apt-get install linux-headers-`uname -r`
cd zaptel
./configure
make
sudo make install
sudo modprobe ztdummy

Install instructions for CentOS as of 3/10/2008 for ztdummy timer only


Install the CentOS kernel sources (google it)
Build the kernel source
Make sure /usr/src/kernel/.config is your kernel config
Make sure /lib/modules/`uname -r`/build points to your kernel directory

Required Packages:
  1. Subversion
  2. ncurses
  3. ncurses-devel
  4. newt
  5. newt-devel
  6. bison
  7. bison-devel

cd /usr/src
svn co http://svn.digium.com/svn/zaptel/tags/1.4.2.1 zaptel
cd zaptel
. ...
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VOIP Billing

March 18, 2015, 4:51 am
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Hosted Billing Services (in Alphabetical Order)


  • Incorpus TeleNetworks Incorpus provides Class 5 and Class 6 softswitches on very affordable monthly rental plans. These are carrier grade switches suitable for Big Enterprises as well as small companies and individuals who are trying to build their own voip company. All switches comes with strong firewalls and bandwidth optimizer and the plans start from as low as just 80$ monthly.Please visit our website and have a live chat with our sales team for any guidance you need. email us for more information at sales@incorpus.in or info@incorpus.in
  • CloudAstrix SPE CloudAstrix SPE is such a VoIP Switch. Build on the world renowned WHMCS Billing Suite, the Soft-switch module brings all necessary functions to perform and provide a top class VoIP service.As a Carrier Neutral soft-switch, CloudAstrix has already proven to be a firm favourite among ISPs all over the world.
    • Note:CloudAstrix SPE Module works with FreeSwitch.
  • Adore VoIP Billing Adore VoIP Billing Software comes with the enhanced functionality along with the architecture with class. It is fully compatible and gets integrated with all other VoIP related products. It is designed with all the present and future demands of booming telecom industry kept in the mind. The telecom industry is changing and developing with rapid speed and the VoIP products such as VoIP Billing comes as an excellent product in this time.
  • 4PSA VoipNow fully featured, carrier-grade, multi-tenant edition for service providers and businesses, that can be installed on their chosen infrastructure or delivered as a UCaaS. VoipNow provides a fast, competitively priced go-to-market solution, from deployment and provisioning all the way to selling and billing.
  • A2BILLING - VoIP Billing Solution / AAA / Class 5 Softswitch.
  • Adore All-in-One SIP Server and Client v2.2.1 - new released with Class5 features
  • Aradial AAA for Billing Solutions
  • benotos offers free callshop billing system 4-level billing system: reseller-subreseller-callshop-customer, 2 different routes, nice easy to use interface, intelligent ratemanager, online payment, detailled reports, receipt printing with own logo, white labelled, use your own brand and domain name and much more features. About 9000 callshops around the world are using our excellent callshop billing solution already. Free signup - best rates on market - low payment amounts
  • BillCall - Telecom Resource Management for wholesale Voip Carriers Panamax’s Telecom Billing Solution BillCall provides solutions for End-User billing, Carrier Access Billing (CABS), CDR Mediation, Rating & Routing.
  • CCM Billing Affordable CDR billing solution for Cisco CallManager telephony systems.
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Asterisk consultants worldwide remote

March 18, 2015, 5:39 am
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freePBX

March 18, 2015, 5:59 am
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FreePBX is a full-featured PBX web application. If you've looked into Asterisk, you know that it doesn't come with any "built in" programming. You can't plug a phone into it and make it work without editing configuration files, writing dialplans, and various messing about.

FreePBX simplifies this by giving you pre-programmed functionality accessible by a user-friendly web interfaces that allows you to have a fully functional PBX pretty much straight away with no programming required. Some of the features that FreePBX supports out of the box are:

  • Unlimited number of Voicemail boxes
  • "Follow Me" functionality
  • Ring Groups with calls confirmation (so if, eg, a cellphone is out of range and diverts to voicemail, all the other phones keep ringing)
  • Unlimited number of Conferences (limited by available CPU power - about 300 simultaneous users in conferences on a P4 3ghz - 600 with a dual core!)
  • Paging and Intercom functionality for man SIP phones that support it.
  • Music on Hold (via MP3s, or streamed off the internet)
  • Call Queues
  • And many other features

FreePBX is built on the LAMPA™ stack (Linux, Apache, MySQL, PHP and Asterisk). It's a modular system, with click-to-install plugins downloadable over the internet from the online module repository.



FreePBX Features at a Glance:



  • Add or change extension and voicemail accounts in seconds
  • Native support of SIP, IAX, and ZAP clients (other endpoints are supported through custom extensions)
  • Supports all Asterisk supported trunk technologies
  • Reduce long distance costs with LCR
  • Route incoming calls based on time-of-day
  • Create interactive Digital Receptionist (IVR) menus
  • Design sophisticated call groups
  • Manage callers with Queues
  • Upload custom on-hold music (MOH)
  • Search company directory, based on first or last name
  • Detect and receive incoming faxes
  • Share administrative duties
  • Backup and Restore your system
  • Save audio recordings of calls
  • View call detail reporting with asterisk-stat
  • View extension and trunk status with Flash Operator Panel
  • View conversation recordings with Asterisk Recording Interface (ARI)

Project Sponsored by SANGOMA

Service Providers

http://sipstation.com

http://www.asterisksiptrunking.us Asterisk SIP Trunking - US — Offers SIP Trunking for Asterisk. Over 500,000 DID's available in 9,500 rate centers. You can activate and setup service in minutes. TDM Enterprise quality. Fully qualified Asterisk consultants ready to assist you. Live customer service with 24/hr ticketing system. No per channel fees, we offer unlimited channels in with your SIP trunk. FREE API for your website. Use our API to leverage the power of our customer user portal on your own website. You can build your own back office admin panel with our API and also provide your customers the ability to order DIDs in REAL TIME along with setting up SIP Trunks. Automate everything and increase customer base.Our Asterisk SIP Trunks are not only for Asterisk. We work with every IP Ready telephone system. Try us out today! No contract and low rates. DIDs are $0.50 per month, Toll Free DIDs are $0.50 per month US/Canada calling rates per minute are $0.009 for TIER1 quality calls. Nothing will route out low quality networks. International calling is available immediately upon registration. Auto refill on your credit card will ensure your account never runs out of funds.

Resources

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VoIP Hardware

March 18, 2015, 6:11 am
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This page lists information about VoIP hardware and VoIP hardware products. For phones and hardware to use with Asterisk, including VoIP phones (both hard and soft phones) and Analog Telephone Adapters, see Asterisk phones.

PSTN Interface cards (analog, GSM, ISDN-PRI and R2/MFC)


This section contains VoIP hardware for connecting analog or digital phone lines from the Public Switched Telephone Network to your Asterisk server. Please keep VoIP hardware providers in alphabetical order.

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2daydirect.com

  • Cisco SPA303 IP phone (3-line business-class IP phone connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX))
  • Cisco SPA504G (For business or home office use full-featured 4-line business-class VoIP phone supporting Power over Ethernet (PoE))
  • Cisco SPA525G2 (Bluetooth enhanced integration with mobile phones to make and receive calls, import your personal contacts, and charge your mobile phone)
  • Cisco SPA514G (Full-featured 4-line business-class VoIP phone supporting Power over Ethernet (PoE) Connects directly to an Internet telephone service provider or to an IP PBX)
  • Cisco SPA508G (Full-featured 8-line business-class IP phone supporting Power over Ethernet (PoE)Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX))
  • Ubiquiti UVP UNIFI VoIP phone (readily integrates with a scalable PBX system for management of features such as call logging, auto-attendant, voicemail, and mass configuration)
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VOIP Phones

March 18, 2015, 7:02 am
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This page is for listing brief details of VoIP Phones including details, where to buy, specifications, and any other relevant VoIP Phone information. Please read the Posting Guidelines for Promoting Products and Services before adding to it.

Hard Phones

Standalone Ethernet Hard Phones (voice only)
An Ethernet hard phone is a self contained IP telephone that looks just like a conventional phone but instead of a conventional phone jack, it has an Ethernet port through which it communicates directly with a VoIP server, VoIP gateway or another VoIP phone. Since a broadband hard phone communicates directly with a VoIP server, VoIP gateway or another VoIP phone it does not require any personal computer nor any software running on a personal computer to make or receive VoIP phone calls. It can be used independently, all that is required is an internet connection. While PC based software solutions are cheaper, a hard phone is the best solution for IP telephony.

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General


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Asterisk security

March 18, 2015, 2:44 pm
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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 - don't depend on it.


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. If you do have remote users, lock your firewall down to only allow those users to connect if possible, rather than opening it to the entire internet. If you have mobile users this may not be an option however.

Other services, such as SSH should be blocked by the hardware firewall. ...
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