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Sat 17 of May, 2008 [14:15 UTC]

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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: 0.45s
  • Memory usage: 2.18MB
  • Database queries: 31
  • GZIP: Disabled
  • Server load: 1.35

PhoneGnome

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PhoneGnome is a free service providing free calls between members and cheap call rates to any other numbers worldwide. PhoneGmome provides four primary means to use the service:

  • Web-Activated (ike Jajah) where you enter the number to call on a website, then PhoneGnome rings your phone and phone number you are calling and connects to the legs together.
  • PC-based - (like Gizmo, FWD, etc) using PC softphone to place calls. The company provides a free Windows-based softphone but the service supports any SIP softphone (including Asterisk)
  • The PhoneGnome box - the company offers a customized ATA/Gatway device that SIP-activates any existing phone service (whether a traditional line, VoIP service like Vonage or Sunrocket, or a Cable phone service). More information is available at PhoneGnome Box
  • Mobile Phone - the company provides a J2ME application that permits use of PhoneGnome service directly on the mobile phone handset. Of course the above web-activated service can be used with moblie phones, as long as the number qualifies as well. An add-on product called MobileGnome ($24.95 per year) provides hop-on/hop-off and callback features. Since the service supports SIP, wi-fi SIP phones can also be used.

Free for calls to members. Rates starting at 2.1c/min for other calls. Users do not need to be on-line to receive calls but they must sign-into their My PhoneGnome portal site periodically to keep their account active.

PhoneGnome members using the PhoneGnome Box can also use any SIP-based service for PSTN terminatrion for calls to non-members (in other words "Bring your own ITSP").

Features:

  • Open Access
    • Buy your PSTN termination minutes from any provider you wish - you are not stuck with only the rates or plans offered by TelEvolution Inc. or any one company
    • Quickly change service, or even use multiple accounts at the same time
    • Buy Virtual numbers (DIDs) from any provider you want
    • We make it easy and convenient to purchase these services from our partners, but you are not required to do so
  • SIP interoperability
    • Open SIP credentials that can be used with any softphone, ATA, Asterisk, Trixbox etc.
    • Exposed SIP addresses permitting directing calls to your PhoneGnome account from any SIP system, Free World Dial, SIPphone, Gizmoproject, Asterisk systems, PBX systems, DID providers etc.
    • SIPbroker support and interoperability - peering with hundreds of VOIP services
    • Use the PhoneGnome box like a remote SIP-based "FXO card" for your Asterisk server
  • Interoperability with IM systems for Voice
    • Free calls with Gtalk, MSN Messenger, and Yahoo! IM users.
    • Optionally connect to Skype on your PC to use your existing regular phone as a Skype phone.
  • Full ENUM support, including e164.arpa and enum.org as well as private ENUM-based VoIP peering
  • Peering relationships representing millions of telephone numbers
  • Support for ITAD/ISN (http://freenum.org/)
  • Free end-user WEB 2.0 XML-RPC APIs with every PhoneGnome account:
    • Interface to contact lists
    • Initiate click-to-dial calls
    • Access call history logs
    • Integrate with your web or blog site
    • Use the APIs on your data enabled mobile phone

See also:

Created by DavidBeckemeyer, Last modification by DavidBeckemeyer on Wed 13 of Jun, 2007 [22:35 UTC]

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