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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
Server Stats
  • Execution time: 0.39s
  • Memory usage: 2.19MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 0.60

Asterisk Features

Asterisk call features



Additional explanations

You may also be interested in this Asterisk feature summary which sometimes provides a slightly less technical description of features. You may find it useful to compare the two sets of explainations as an aid to understanding.



ADSI On-Screen Menu System :

Alarm Receiver : Receive a alarm on your phone.

Append Message: Append the message to an e-mail.

Blacklists: Screening malicious callers.

Blind Transfer: Blindly transfers a call.

Call Monitoring:

Call Parking: Call park allows you to place a call on hold on an extension other than your own. This is useful for retail stores or factory floors.

Call Queuing: Queuing up the call.

Call Recording: Recording conversations.

Call Retrieval: Paging for the correct person to pick up the call.

Call Snooping:

Call Transfer: Transfering a call to another extension.

Call Waiting: Allows the use of a hook-flash to change between two simultaneous calls.

Calling Cards:

Conference Bridging:

Database Store / Retrieve:

Database Integration:

Dial by Name: Assuming voicemail is set up correctly, Dial by Name allows an outside caller to get 411-like help in finding the extension number of the person they wish to call.

Direct Inward System Access: Allows an outside caller to have full access to PBX functions, which is normally reserved for internal-only access. Common use of DISA is if you have road warriors or home-based workers: they can call into your PBX and make free outbound long distance calls, also appearing to be at your physical location.

Distinctive Ring: The called phone has distinct ring based on the place from where call originated eg., a specific number, internal or extenal calls.

Distributed Universal Number Discovery (DUNDiâ„¢) :

Do Not Disturb: No calls will be connected.

E911: Enchanced 911, important in case of a VoIP solution.

ENUM: Universal number for your VoIP phone.

Fax Transmit and Receive (3rd Party OSS Package) :

Flexible Extension Logic:

Interactive Directory Listing:

Local and Remote Call Agents: People can log on to the PBX system
from any phone using a Login ID, allowing them to receive and respond to calls.

Macros: In configuring the dial plan, the use of macros and pattern matching greatly simplifies identical configuration for hundreds, even thousands, of extensions.

Predictive Dialer: Generally for use in outbound telesales or customer service; predictive dialing optimizes the time that agents spend on the phone by pre-dialing numbers when it is determined that an agent will be coming available.

Privacy:

Open Settlement Protocol (OSP) : OSP is a protocol standard for securely routing and accounting for inter-domain VoIP calls. OSP is not a VoIP protocol, but rather a standard for managing the billable exchange of VoIP sessions between IP networks. OSP is used by wholesale VoIP carriers to establish secure point to point peering between source and destination networks. The details for OSP, officially known as "ETSI TS 101 321 Open Settlement Protocol (OSP) for Inter-Domain pricing, authorization and usage exchange", can be found at www.etsi.org.

Overhead Paging: Sometimes also called intercom, this allows a centrally-located speaker to be "dialed into", for making announcements. Frequently seen in retail stores, also car dealers, factory floors and other non-office type situations.

Protocol Conversion: Can bridge calls between dissimilar systems, for example IAX to SIP, PSTN to MGCP.

Remote Office Support:

Roaming Extensions:

SMS Messaging:

Spell / Say: Can "read" letters and numbers to a caller, for example reading back the current time.

Streaming Media Access: Allows streaming of media such as mp3s directly into the phone system, useful for hold messages.

Supervised Transfer: Similar to blind transfer, allows a person to transfer a call to another extension by first announcing that call to the transfered extension. Useful in situations like "I have Joe Carseller on the phone, do you want to talk to him?"

Talk Detection: There are a few modules which allow various types of noise detection on certain channels; talk detection can be used to trigger an event, for example, play an automated message when somebody says "hello."

Text-to-Speech (via Festival) : Asterisk may read text via the Festival voice synthesis suite. For example, this could be used to read a web page containing the current weather or an email

Three-way Calling: Adding a third party to an ongoing conversation.

Time and Date:

Transcoding: Allows Asterisk to bridge calls that use different audio encodings, for example g711.u to GSM.

Trunking: Logically grouping together multiple phone lines for outbound dialing; typically seen in medium to large deployments. The PBX can be configured to auto-select an available line for outbound dialing rather than the user having to choose which line to dial out on.

VoIP Gateways: Asterisk can act as a bridge between VoIP telephones and the PSTN; additionally, Asterisk can be used to route calls to some third-party VoIP gateways (think Vonage, though I don't believe that is an available option). Check out BroadVoice or Packet8.


Voicemail to email: Asterisk's native voicemail can send an email to the voicemail recipient, and can optionally attach a WAV file of the entire message.

Voicemail Groups:

Web Voicemail Interface: As with most things Asterisk, it is almost trivial to provide a browser-based interface to the system.


Zapateller: Built-in, automated "black hole" for inbound telemarketing calls.


Pease feel free to add as many definitions as possible
Thanks to http://forum.digium.org for their support
Created by ashutosh, Last modification by Christoph Reichl on Thu 09 of Nov, 2006 [16:41 UTC]

Comments Filter

by Olivier on Tuesday 20 of November, 2007 [15:48:56 UTC]
I don't think we can say "Dial by Name: Assuming voicemail is set up correctly, Dial by Name allows an outside caller to get 411-like help in finding the extension number of the person they wish to call".

Either this feature is not supported (Automatic Speech Recognition) or this definition should be improved.
Dial by Name seems a natural hardphone feature, not a B2BUA feature.

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