Voip Networks/Free Hosted PBX Systems

Voip Networks/Free Hosted PBX Systems

Voip Networks/Free Hosted PBX Systems

1.VoXaLot: (www.voxalot.com)

VoXalot can be best described as a service aggregator. In a few words they let you put multiple services together, to create a custom VSP. They are not just a PBX, but rather a network on their own...

Free VoXBasic Service includes:
  • Incoming calls from SipBroker Access Numbers
  • Incoming calls via SIP URI
  • Incoming calls from DIDs forwarded to a SIP URI
  • Multiple Outgoing Providers can be registered (this supports most providers, except those that require SIP registration for placing calls)
  • Dial Plans or Smart Calling which lets you use the cheapest provider for each destination
  • Speed Dial List
  • Voice Mail (which can also be accessed from regular phones)
  • Automatic ENum lookup
  • Automatic Geo lookup for premiumnumbers in the UKand Australia (via e164.org)
  • Virtual Toll Free (A small script to let people call you from the Web: VoXalot can call their phones, at your expense, or their SIP URI and connect them to you)
  • Import/Export feature for Providers/DialPlans/Speed Dials to move them form one acct to the next
  • Can be used with any SoftPhone, ATA, and a number of VoIP enabled mobiles


VoXPremium adds the following extra features:
  • Call Forwarding based on Caller ID or Provider and Time of Day(send incoming calls to SIP URIs or Phone numbers, so you can always receive your calls)
  • Web CallBack (enter two numbers and the providers to use for each number and place a call, hence using VoXalot form PSTN phones as well)
  • VoiceMail on the Web
  • Incoming Calls from third party providers via SIP Registration (VoXLite: 1, VoXPro: 5, VoXExtreme: 10)

They also provide the SIPBroker service with 200+ donated access numbers around the world to call Voxalot and over 2000+ other SIP based VoIP Networks (millions of VoIP users) for free...

They are associated with (but do not run) the only ENum service open to the end-user, e164.org

2.PBXes(www.pbxes.com)

They provide a hosted Asterisk based PBX service. Their interface remains faithful to the Asterisk interface in general, although it is much more user friendly. Their services include:

PBXes Free: The Free account notable limits are a 10 GB usage, and 2 simultaneous calls, with a 60 min cutoff, and the unability to post in the support forums.
PBXes Premium: Includes a full on PBX including among others Fax as EMail,a full AutoAttendant and Que system, and CallBack access to your Trunks
PBXes Pro: Is geared towards businesses, providing a hosted PBX which can be customized for various clients, including much of the same features as the Premium acct.

For a breakdown and description see: https://www4.pbxes.com/iptel_details_e.html

3.SipSorcery (www.sipsorcery.com)

MySipSwitch is a PBX service concerned solely with Call Routing. In avoiding the hadling of the actual audio stream their application runs much lighter. It is currently an R&D base being run by two technicians and sponsored by BlueFace.ie It provides the following fetaures:
  • Multiple outgoing SIP Providers
  • Outgoing Dial Plans
  • Multiple Incoming SIP Providers
  • SIP Registration unto a third party destination URI
  • A Call Routing facility handled via DIAL PLANS that has lately added support for multiple simultaneous call forwarding


Their service is new, and the interface is a bit raw, but their outlook looks promising, and used in conjuction with other PBX services, can contribute to a solid VoIP arrangement...

For more info see: http://www.mysipswitch.com/forum/viewtopic.php?t=139


4. LiberaILVoip (www.LiberaIlVoip.it)

Their website is in italian. They provide the following services:

  • Incoming and Outgoing Provider Registration
  • Dialing Rules
  • Incoming Call Filters and Call Forwarding
  • VoiceMail
  • Gateway Access (CallThrough)
  • Call Back access to your provider rates (the only PBX service to offer this for free to my knowledge)
*Some limitations apply, will update when I have a better understanding of what they are


5. GTalk2Voip (http://www.gtalk2voip.com/)
-Used in conjuction with MSN, Yahoo, GoogleTalk messengers
-You create an account by simply entering your email in the home page
-service@gtalk2voip.com (or something similar) adds you as a friend.
-Open a Convo and type: MYPAGE
-Click on the link for your account page
-To CALL type commands on convo with service@gtalk2voip.com :
SIP URI:
Code:
CALL user@provider.com
MSN:
If email is username@hotmail.com
then type
Code:
CALL username_at_hotmail.com@msn.gtalk2voip.com
If email is username@msn.com
then type
Code:
CALL username_at_msn.com@msn.gtalk2voip.com
GoogleTalk:
If email is username@gmail.com
then type:
Code:
CALL username_at_gmail.com@gtalk.gtalk2voip.com
If email is username@DomainWithGoogleApps.com
then type:
Code:
CALL username_at_DomainWithGoogleApps.com@gtalk.gtalk2voip.com
Yahoo
If email username@yahoo.com
then type:
Code:
CALL username_at_yahoo.com@yahoo.gtalk2voip.com
PSTN
Type:
Code:
CALL 14161112323
(always type number in international format)
PSTN calls are not free, but you can now add your own providers to GTalk2Voip, and the call will be routed through them. Otherwise GTalk2Voip routes through its own providers and requires you have a balance
-When calling someone for the first time using the commands above, chances are it will not go through, but they will receive a notice that service@gtalk2voip.com has added them as a friend, if they accept, the second time you try they will receive a call from service@gtalk2voip.com
-This works much smoother in GoogleTalk, rather than MSN (have not tested with Yahoo)
-They assign you a number *018xxxxx, that can be reached by dialing SipBroker Access Number + *018xxxxx
or *018xxxxx@sipbroker.com (SIP URI calling), in all this cases you'll receive a call on your MSN/Yahoo/GTalk messenger
-You can also get a DID number, forward it to *018xxxxx@sipbroker.com, and your MSN/GTALK/YAHOO messenger will ring when someone calls you
-There is a generic Voicemail if you don't pick up, or you can forward to a SIP URI and receive calls to your current provider
-You can use them to IM people not using same messenger (http://www.gtalk2voip.com/gtalk_service_im.shtml)
-You can create your own WebCall button (it calls you via Voip, plus uses one of your providers to call the other party and connect you together, if someone enter an actual number though, you will end up paying for the call, but at VoIP rates) (http://www.gtalk2voip.com/webcall.shtml)
-In Summary, neat to play around with. It is a raw framework for now, not very user friendly, but looks promising.
-Using GTalk2Voip with your own Domain: http://www.gtalk2voip.com/forum/topic_show.pl?tid=41

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