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Trixbox Asterisknow Comparison Essay

Created by: jht2, Last modification: Wed 09 of Aug, 2017 (19:40 UTC) by freevoice

Asterisk

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Asterisk is a complete PBX in software. It runs on Linux, BSD, Windows (emulated) and OS X and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny. Check the Features section for a more complete list.

Asterisk needs no additional hardware for Voice-over-IP, although it does expect a non-standard driver that implements dummy hardware as a non-portable timing mechanism (for certain applications such as conferencing). A single (or multiple) VOIP provider(s) can be used for outgoing and/or incoming calls (outgoing and incoming calls can be handled through entirely different VOIP and/or telco providers)

For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsor, Digium. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks . In addition, single to quad port analog FXO and FXS cards are available and are popular for small installations. Other vendors' cards can be used for BRI (ISDN2) or quad- and octo- port BRI based upon CAPI compatible cards or HFC chipset cards.

For interconnection with the cellular network (GSM or CDMA), Asterisk can use the Celliax channel driver or chan_mobile that is in the trunk now and there is also a unofficial backported version.

Lastly, standalone devices are available to do a wide range of tasks including providing fxo and fxs ports that simply plug into the LAN and register to Asterisk as an available device.

If you are looking for a software-based IP-PBX for Windows, 3CX offers a highly-regarded product.

The current release versions of Asterisk are 1.2.37, 1.4.36, 1.6.0.28, 1.6.1.20. 1.6.2.13, and Asterisk 1.8.0. The current beta is Asterisk 1.8 beta 5.

This Wiki covers both the stable and the development branch of Asterisk. When adding new commands, applications and options, please also add a note on *when* this was added so that users may compare with their version date.

Know All About Asterisk VoIP

November 29, 2016 –There are times when you would need to have Asterisk VoIP services for your company. This type of service has much better quality and reliability than traditional VoIP companies that are
VoIP providers that have options for call centers and businesses. You can find some great options for your required services without having to settle for minimum payments or any set monthly fees.


Daily Asterisk News




VoIP Today News




Other Asterisk News

  • 2015-05-02 - DMLink releases their 8 ports GSM asterisk card.
  • 2015-05-02 - DMLink releases their 8 ports E1 asterisk card.
  • 2015-02-07 - 2-Way Radio over IP Solution for Asterisk
  • 2014-09-04 - OpenSIPS Summit Las Vegas 2014 registration is officially open! Come experience a rich collection of workshops about OpenSIPS and Asterisk integration
  • 2014-03-25 - Sequoia Mobile announces "alpha" version of Asterisk connector for Nuance ASR Recognizer
  • 2014-03-13 - Agile CRM announces VoIP telephony integration. Source: PRWeb
  • 2013-10-18 - Asterlook releases the Asterlook software that manages calls in Outlook with Asterisk IPBX; Press Release
  • 2013-04-30 - Netfors releases chan_ss7 2.2.0 for Asterisk 11.x.
  • 2013-03-28 - Asterisk 11.3 Asterisk 11.3 is the new current release.
  • 2013-03-15 - DMLink releases their 24 port PCI-E asterisk analog card , runs DAHDI standard driver.
  • 2013-03-15 - DMLink releases their single port PCI-E PRI asterisk card , runs DAHDI standard driver.
  • 2013-03-05 - Exelysis Contact Center - The new version of Exelysis Contact Center for Asterisk is now released.
  • 2012-10-20 - Asterisk compatible VXML server IEC - SIP/2.0+VXML 2.1 IVR Platform SRGS MRCP Server, Cloud/MRCP ASR/TTS , Linux/Windows Free One Channel.
  • 2012-10-12 - Intuitive Voice Technology launches new reseller offering get three FREE PBX licenses for signing up.
  • 2012:09:11 - Version 3.2 firmware for Zycoo ZX50 Series IP-PBX released Change log
  • 2012-08-30 - DMLink releases their 4 port PCI-E asterisk card , runs DAHDI standard driver.
  • 2012-04-12 - WeQ4U releases iPhone app that interfaces with their asterisk based call center queueing system.
  • 2012-04-11 - DMLink releases their 8 port GSM asterisk PCI card GSM800P, runs DAHDI standard driver.
  • 2012-04-11 - DMLink releases their new 24 port analog asterisk PCI card, runs DAHDI standard driver.
  • 2012-03-28 - DMLink releases their online shop 2.
  • 2012-03-28 - DMLink releases their online shop 1.
  • 2012-03-08 - DMLink releases their 16 port PCI-E analog asterisk PCI card, runs DAHDI standard driver.
  • 2012-03-07 - OpenVox Delivers The Cost Effective Tapping Application
  • 2012-02-24 - OpenVox Releases Chan-Extra 2.0.1 To Add More Functions To GSM Applications
  • 2012-02-14 - OpenVox Released Multiform Transcoding Solutions
  • 2012-01-16 - DMLink releases their new 4 port analog asterisk PCI card, it runs DAHDI standard driver.
  • 2012-01-10 - WeQ4U releases asterisk based call center queueing system. WeQ4U.
  • 2011-12-27 - OpenVox Launches A Complete Asterisk-Based Appliance
  • 2011-12-14 - Humbug And OpenVox Partner To Prevent Telecom Fraud
  • 2011-12-09 - OpenVox Distributes New Octasic® SoftEcho Supporting DAHDI Driver
  • 2011-11-29 - DMLink releases their new 16 port analog asterisk PCI card, support DAHDI standard driver, no patch needed.
  • 2011-11-18 - OpenVox Announces FREE 24*7 Technical Support Program For Open Source Community
  • 2011-11-15 - OpenVox Announces BRI Products Price Reduction with A Lifetime Warranty Upgrade
  • 2011-08-26 - OpenVox New Embedded Asterisk Motherboard Released
  • 2011-08-01 - PowerAST The World first open hardware/software designed embedded PowerPC dual core 1.2G CPU IPPBX platform (Up to 16 E1/T1/J1 with hardware EC modules supported) has been released first version.
  • 2011-07-29 - Intuittech-Blog Over 600 people strong, in 7 countries. The worlds largest Asterisk organization takes shape!
  • 2011-07-20 - VadaXchange Buddy version 1.0 released - new user-friendly call control interface developed on top of an Asterisk PABX. About time business PABX become user friendly!
  • 2011-06-27 - Atcom PRI Cards New range launched - AX-1D, AX-2D and AX-4D. Optional Hardware Echo Cancellation module available.
  • 2011-05-16 - Orderly Stats version 1.8 of OrderlyStats for Asterisk Call Centers has been released
  • 2011-05-11 - Positron G1000 Asterisk Appliance Released - 150 concurrent calls
  • 2011-05-10 - Positron Telecom 1U Rackmount PBX is now available
  • 2011-04-22 - The Alvis-4-Ext: T1/E1 External PRI Channel Bank based on RtpBridge solution is announced by Odin TeleSystems!
  • 2011-04-19 - DMLink releases their new 4 port GSM asterisk PCI card.
  • 2011-04-09 - Free Asterisk Technology Day in Kuala Lumpur, 28 April 2011
  • 2011-03-30 - How to add a Flash softphone to your webpage, if you use Asterisk PBX
  • 2011-03-17 - Microbase CTI Microbase announced the release of Microbase CTI v2.0 - The pure Asterisk Based CTI Solution.
  • 2011-03-02 - OpenVox Releases Hardware Echo Canceller for ISDN BRI Cards with LIFETIME warranty.
  • 2011-02-24 - Evolution PBX 4.0 released (100's of new features) by Intuitive Voice Technology
  • 2011-02-24 - DMLink releases their new 16 port analog asterisk PCI card.
  • 2011-02-15 - Netfors has released version 2.0.0 of the SS7 channel driver for Asterisk, chan_ss7: Download here.
  • 2011-02-02 - Intuittech Sdn Bhd launches Nagmon for Asterisk the monitoring appliance in conjunction with the DIGIUM ASTERISK WORLD 2011; Press Release
  • 2011-02-02 - Membantu24 the first global Asterisk support centre is launched in conjunction with the DIGIUM ASTERISK WORLD 2011; Press Release
  • 2011-02-02 - Hayibo! the softphone for the Enterprise releases the free version of its softphone in conjunction with the ITEXPO EAST 2011 in Miami; Press Release
  • 2011-01-07 - Asterisk Nortel Asterisk Avaya Communications Manager with SES ,Avaya IP Office R6.0 ,BroadSoft BroadWorks


Starting Out


Books





Introduction

  • Asterisk introduction: An overview for new Asterisk administrators - THE PLACE TO START!!
  • An introduction to Asterisk, The Open Source Telephony Project - Basic setup How-To/tutorial, installation, SIP-devices and dialplan, with test applications.
  • Asterisk: A Non-Technical Review (pdf): An overview for executives and managers
  • Asteriskguru Tutorials A huge collection of tutorials for asterisk.
  • Asterisk installation
  • Asterisk IRC logs: #asterisk IRC logs
  • Asterisk Mailing Lists
  • Asterisk software addons
  • Asterisk tips and tricks: Solutions to common problems, hints of what you can do with this powerful software
  • Asterisk video training Asterisk and Linux step by step installation guide
  • Asterisk video training Free videos on Asterisk, trixbox, and FreePBX
  • Asterisk video training Free videos on Asterisk, trixbox, and FreePBX
  • Series of video tutorials about Asterisk dial plan development
  • Blindhog.net - Video Tutorials.
  • Collection of tutorials for Asterisk dial plan development
  • Development Which development environment is best for my voice app? (Asterisk, usually!)
  • FAQ and SEARCH helper: Look for answer for the question that bugs you here!
  • Free Introduction to Asterisk eBookcomplete setup tutorial
  • Free video tutorials about Asterisk dial plan
  • http://www.suvi.org/theory/asterisk.html Gives a good German introduction howto setup Asterisk quickly.
  • Linux 101: A beginners guide to using Linux
  • Linux 101: A small wiki with snippets of useful Linux info
  • News, Project status and roadmap
  • Systm 5 - Asterisk: Video @ YouTube
  • VoIP User Groups: Local resources in your area
  • Where to download Asterisk

Asterisk based Solution

  • Automated Appointment Reminder System to send automatic reminders of appointments to yourself, staff, clients, prospects and other business entities via call, SMS, fax, email
  • Automated Ticketing IVR System to book tickets of flight, bus, train, movie, etc. using an interactive voice response system.
  • Bicom Systems PBXware is an IP PBX turnkey business communications platform that is flexible, reliable, and scalable. PBXware was the first ever GUI for Asterisk, presented at Astricon in 2004, setting the standard for others to follow. Available in Business, Multi-Tenant, and Call Center Editions.
  • CRM Integrated IVR System is an effective tool to cater your customers in a personalized manner.

Hardware



Administration and system layout


Configuration


Management

  • Asterisk CLI: The interactive command prompt language
  • Asterisk GUI: Web and other interfaces to Asterisk for management and configuration
  • Asterisk Manager API: The Asterisk Manager API
  • Asterisk options: Command line switches when you start your Asterisk PBX
  • Asterisk Zeroconf Support: Service Discovery for Asterisk using Zeroconf
  • DooxSwitch.com: Asterisk based advanced managment and communication solution. Routing, Billing, PBX, IVR menu, Callshop, Callcenter, DID shop , SMS, Calling Cards , Dialers , Softphones flexible API and many more.
  • Asynchronous Javascript Asterisk Manager (AJAM) - HTTP Manager API Access

Troubleshooting


General Reference


Country-Specific Information


Commercial support


SIP Service Providers

DIDForSale SIP Trunking - DIDForSale provides largest coverage over US, UK and Canada. Offer Unlimited SIP Trunking Service for businesses.
Cebod Telecom provides SIP trunks, Virtual PBX telephony solutions, internet services for small to mid-sized enterprises, and larger companies delivering high- quality voice, premium features and unified communications. The company offers a state-of-the-art Pay-Per-Line business phone service model, empowering businesses to pay per line as opposed to paying per user in the company. In addition offers all-inclusive premium IP PBX and dial tone features including auto attendant, call tracking, call logs, music on hold, call forwarding, voicemail, call queue, conference bridge, e-Fax, and SMS and more. High call quality, unlimited user extensions, and unlimited calling within USA and Canada are some of the benefits also offered for each phone plan. Contact us today for a free demo.
  • DIDLive provides Asterisk SIP Inbound Service. Local DID Numbers in USA/Canada & 40 other countries. USA/Canada Toll Free Numbers.
Contact DIDLive at 1-212-901-0800 or visit us at www.didlive.com

User Groups



Weekly SIP Asterisk Users Conference

  • x2z.eu http://www.x2z.eu/ THis conference is open to users at all levels of asterisk expertise

Interesting and Unusual Projects

  • Village Telco Using Asterisk in a WiFi Mesh network to provide phone service in rural areas

Howtos and Tutorials


Asterisk Admin GUI v13

For Asterisk Admin Gui version 2.11 setup information please click here
For Asterisk Admin Gui version 12 setup information please click here


GENERAL INFORMATION
Asterisk Admin GUI is an open source interface for configuring the Asterisk PBX server. The Asterisk Admin GUI interface can vary slightly depending on which distribution you use. This may be directly from the Asterisk Admin GUI website or through one of the major Asterisk distributions such as trixbox, Elastix, PBX in a Flash, etc...

With Asterisk Admin GUI you are able to configure most of Asterisk's options without editing the individual configuration files. You can also setup advanced options such as call routing, voicemail, and other calling features in a more manageable interface. Below we provide some resources you can visit to obtain further information.

Please note that Callcentric is not responsible for preventing unwanted physical or remote access your IP PBX. If your IP PBX is compromised then you will be responsible for any damage caused.

Please be sure to read this guide regarding securing your IP PBX solution.


RESOURCES
Websites
Elastix
PBX-in-a-Flash

Help / Support
Asterisk entry via Voip-info
Elastix support
PBX-in-a-Flash support


Configuring Asterisk PBX (chan_sip) using the Asterisk Admin GUI interface
Here we will configure Asterisk through the Asterisk Admin GUI administrative interface to properly route both incoming and outgoing calls to and from Callcentric. This guide assumes that you have installed Asterisk Admin GUI using either the Asterisk Admin GUI package (or distro), Elastix, IncrediblePBX or a method of your choice. This guide also assumes that the Asterisk Admin GUI install steps were completed properly and that you have administrative access to the Asterisk Admin GUI administration interface.

This guide is based on version 13.0.113 of the Asterisk Admin GUI (running Asterisk 13.7.1)

We recommend that you read each step through in its entirety before performing the action(s) indicated in the step.

We also recommend that you check which version of Asterisk your PBX is based on; as there are many significant changes between each revision of Asterisk. To check which version your PBX is based on; please log into your PBX's command line interface and execute the command "show version" or "core show version", and you should see an output similar to the following:



STEP 1Trunk Configuration
In the context of this guide a trunk is used to route calls between your Asterisk PBX and your desired VSP (Voice Service Provider), in this case Callcentric. To configure a SIP Trunk, please proceed with the following:

  1. Login to Asterisk Admin GUI administrative interface

  2. From the navigation bar at the top of the page, click on Connectivity >> Trunks

  3. Click the Add Trunk button that is located in the middle of page, and select Add SIP (chan_sip) Trunk from the drop down menu

  4. In the General section, locate the Trunk Name option, and specify callcentric in the given field

  5. Click on the SIP Settings tab, and click on the Outgoing sub-section tab

  6. Locate the Trunk Name option, and specify callcentric in the given field

  7. Copy and paste the following into the PEER Details field.

    context=from-pstn-toheader fromdomain=callcentric.com fromuser=1777MYCCID host=callcentric.com insecure=port,invite secret=SUPERSECRET type=peer defaultuser=1777MYCCID disallowed_methods=UPDATE directmedia=no videosupport=no disallow=all allow=ulaw

  8. Click on the Incoming sub-section tab

  9. Enter your Register string in this format:

    1777MYCCID:SUPERSECRET@callcentric.com

    *** Where 1777XXXXXXX is either the default extension 1777MYCCID OR 1777MYCCIDEXT, where 1777MYCCID is the 1777 number assigned to you by Callcentric and EXT is the three digit extension you are trying to register this UA to.

    *** Where SUPERSECRET is your extension SIP Password here. Your extension SIP password is the password you created for the extension you are trying to use. You may edit the SIP password you wish to use in by logging into your My Callcentric account and clicking on the Extension menu link and then modifying the appropriate extension.

  10. Click on Submit Changes to add your new SIP trunk to your Asterisk server

  11. Click on the Apply Config button at the top of the screen to apply the changes you've just made







  12. From the navigation menu, click on Settings >> Asterisk SIP Settings

  13. From the sub section General SIP Settings, locate the option Allow Anonymous Inbound SIP Calls, and set this option to No

  14. Click on Submit Changes to save your changes

  15. Click on the Apply Config button at the top of the screen to apply the changes you've just made



  16. From the sub-menu bar, click on Chan SIP Settings tab

  17. Locate the option Allow SIP Guests, and set this option to No

  18. Locate the option SRV Lookup, and set this option to Yes

  19. Locate the option Other SIP Settings, and use the following settings:

    sendrpid = yes trustrpid = no disallowed_methods = UPDATE

  20. Click on Submit Changes to save your changes

  21. Click on the Apply Config button at the top of the screen to apply the changes you've just made



  22. From the navigation bar at the top, visit the Admin >> Config Editor (this assumes that you've installed the Config Editor module on your Asterisk Admin GUI installation. If you have not installed it, you can install it by visiting the Admin >> Module Admin configuration page)

  23. On the left pane, locate the file labeled sip_custom_post.conf and copy and paste the following into that file:

    [callcentric1](callcentric); host=alpha1.callcentric.com [callcentric2](callcentric); host=alpha2.callcentric.com [callcentric3](callcentric); host=alpha3.callcentric.com [callcentric4](callcentric); host=alpha4.callcentric.com [callcentric5](callcentric); host=alpha5.callcentric.com [callcentric6](callcentric); host=alpha6.callcentric.com [callcentric7](callcentric); host=alpha7.callcentric.com [callcentric8](callcentric); host=alpha8.callcentric.com [callcentric9](callcentric); host=alpha9.callcentric.com [callcentric10](callcentric); host=alpha10.callcentric.com [callcentric11](callcentric); host=alpha11.callcentric.com [callcentric12](callcentric); host=alpha12.callcentric.com [callcentric13](callcentric); host=alpha13.callcentric.com [callcentric14](callcentric); host=alpha14.callcentric.com [callcentric15](callcentric); host=alpha15.callcentric.com [callcentric16](callcentric); host=alpha16.callcentric.com [callcentric17](callcentric); host=alpha17.callcentric.com [callcentric18](callcentric); host=alpha18.callcentric.com [callcentric19](callcentric); host=alpha19.callcentric.com [callcentric20](callcentric); host=alpha20.callcentric.com

    ***** The value that you've specified in the option Outgoing settings >> Trunk Name within your trunk configuration page, must be specified within the parentheses on the settings above; so if you've used "callcentric" (all lowercase) for your Trunk Name (as presented in our example), please specify the above settings as-is. Or for example if you've specified Trunk_1 for that option, you will need to specify the following:

    [callcentric1](Trunk_1); host=alpha1.callcentric.com [callcentric2](Trunk_1); host=alpha2.callcentric.com etc...

    Your configurations should look similar to the screenshot below:



  24. Click on Save to save the file above

  25. Click on the red button labeled Apply Config at the top of the screen to apply the changes you just made

* If you do not have the Config Editor module installed, you will need to log in to your server and edit the /etc/asterisk/sip_custom_post.conf file manually, usually with an editor such as nano.

STEP 2Outbound Route Configuration
An outbound route sends calls which are dialed in a certain pattern to your desired provider, in this case Callcentric.

  1. From the navigation bar, click on Connectivity >> Outbound Routes, to configure your PBX to route outgoing calls towards your Callcentric trunk

  2. From the sub-section Route settings, enter to-callcentric into the Route Name field

  3. Locate the Trunk Sequence for Matched Routes section, and select the callcentric trunk from the drop down list



  4. From the sub-menu bar, click on Dial Patterns

  5. Locate the Dial Patterns that will use this Route section, and specify the following options:

    prepend
    prefix9
    match pattern.
    CallerID


  6. Click on Submit Changes to add your new route to your Asterisk server

  7. Click on the Apply Config button at the top of the screen to apply the changes you've just made



STEP 3Extension Configuration
In this step, we'll create a local extension on your PBX. This local extension (on your PBX) provides an account number that another User Agent (software or hardware used for calling) can connect to in order to make and receive calls. There are a few types of extensions; here we will create a SIP Extension.

If you have already configured an extension then you may skip this step. Then in the next step (Inbound Route Configuration) you may use your pre-configured extension.

  1. From the navigation bar, and click on Applications >> Extensions to add a new extension which will connect to your Asterisk server

  2. From the drop-down menu, select Generic SIP device (or Add New Chan_SIP extension)

  3. Enter 1000 as the User Extension

  4. For now we will use a generic identifier for this extension. Enter First Extension for the Display Name field. Later you may enter a unique identifier of your choice

  5. Enter your desired password in the Secret field. You will use this password when configuring your desired UA later in order to connect to your Asterisk PBX

  6. Click on Submit Changes to add your new extension to your Asterisk server

  7. Click on the Apply Config button at the top of the screen to apply the changes you've just made



STEP 4Inbound Route Configuration
With an inbound route you are given the flexibility to send incoming calls to a wide range of destinations. For example you may route an incoming call to a specific extension, to a ring group, or to an IVR. In this section we are going to setup an inbound route which will handle ANY incoming call on ANY numbers, including emergency numbers, and simply route those calls to a specific extension (1000). Later on you can configure more complex routing schemes, such as DID Based Routing.

If you have already configured an extension then you may substitute your pre-configured extension for point 4 below.

  1. From the navigation bar, click on Connectivity >> Inbound Routes to configure the routing of calls to your Callcentric account.

  2. If there isn't a default inbound route defined already within your PBX, then click on Add Incoming Route. You will first want to fill the DID Number field with your 1777 number, or if you've acquired a phone number from us already, please use your phone number in the DID Number field (i.e. if you've acquired the number 12125551000, please use 12125551000 on this field). Make sure to leave the Caller ID Number blank in order to match any incoming call. This is useful if you wish to receive all calls

  3. Scroll down to the Set Destination section

  4. Choose First Extension (1000) from the Core drop down box

  5. Click on Submit Changes to add your new inbound route to your Asterisk server

  6. Click on the Apply Config button at the top of the screen to apply the changes you've just made





STEP 5Configure and test UA
  1. Choose your desired UA

  2. Use the IP address or hostname for your Asterisk box along with 1000 (the extension created earlier) and password for the 1000 extension to connect to your Asterisk box

  3. Next you will want to try placing test calls to and from your Asterisk PBX using the UA currently connected to your newly created extension (1000)

STEP 6Placing Test Calls
You can make a test call to 17771234567, or if you are signed up for one of Callcentric's rate plans you can place a call to a traditional landline or mobile phone by dialing either:
1 + the area code and number for calls to the US
Or
011 + the country code, area code, and number for calls worldwide (you may also use 00 instead of 011).

To test inbound calls from Callcentric to your Asterisk installation, follow the directions listed in this FAQ.

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