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Patton

January 10, 2013, 7:02 am
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About Patton


Patton manufactures electronic communications equipment for carrier, enterprise, and industrial networks worldwide. Incorporated in 1984.
the Patton catalog includes over than 1000 products.
  • SmartNode VoIP Gateways, Routers, Integrated Access Devices (IADs), and Enterprise Session Border Routers (ESBRs)
  • ForeFront multi-service access infrastructure solutions (T1/E1, G.SHDSL, xDSL, dial-up)
  • IPLink CPE solutions (WAN routers, modems, remote access servers, NTUs, CSU/DSUs)
  • CopperLink Ethernet Extenders
  • EtherBITSdevice servers
  • EnviroNET industrial-networking equipment, network-connectivity
  • SerialComm device networking solutions (interface-converters, etc.)
  • and more

Patton is headquartered in Gaithersburg, MD, USA with branch offices around the world, in including Bern, Beirut, and Budapest. For more information or a free catalog, contact sales@patton.com.

Patton Electronics Company
7622 Rickenbacker Drive
Gaithersburg, MD 20879
USA
Tel: (301) 975-1000
Fax: (301) 869-9293
Email: sales@patton.com

Web: www.patton.com

Where to buy:

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Toll Free

January 11, 2013, 8:21 am
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To receive calls to USA and Canada TollFree numbers (800, 866, 877, 888, 855), you will need and Toll Free Number service provider.
To send calls to USA and Canada TollFree numbers you can either send these calls to your usual long distance service provider (who may charge you by the minute) or send the calls to one of these free: Toll Free Termination Providers. If you have a large volume of calls to tollfree numbers, some Toll Free Termination Providers will pay you for the calls.

Toll free (800, 866, 877, 888, 855) phone number providers:

(In alphabetical order):
  • Alcazar Networks Inc is offering Wholesale VoIP Origination / Termination / Free Toll Free Termination - Get paid for your toll free traffic! Wholesale SIP
  • Anveo offers USA, Canadian and international Toll-Free phone numbers. Anveo provides web based Visual Call Flow Designer where you can easily create custom IVR Call Flow for your business. Anveo has many exceptional features including click to call, ringing multiple phones, call transfers to Skype or Gizmo5, conference calls, online FAX, voicemail, extensions, Business Control Panel to create and manage employee's sub accounts and more!
  • BinFone Telecom offers toll free service, delivered via SIP and IAX.
  • ComCanada CommunicationsCRTC Registered, Provider of Hosted PBX, DID service, Toll Free, equipment sales, consulting, and retail/wholesale origination & termination. Supported Protocols: SIP & IAX 1-877-697-VOIP
  • Database Systems Corp.Toll Free Phone Services - IVR toll free services. Products include voice broadcasting, IVR systems, IVR software and IVR hosting services.
  • Easy Office Phone - Toll free services delivered via SIP or with our Hosted PBX platform.
  • Flowroute LLCWholesale VoIP, A-Z SIP Termination, Cheap DIDs, Toll-Free Origination, Free CNAM Storage, E911, T.38 Fax support, SIP-TCP support, Redundant servers, Graph Statistics and great interface
  • FonAngle Hosted PBX - Offers Toll free services (including Vanity orders) in the United States, Canada and select International destinations via our Hosted PBX or SIP Trunking platform.
  • Fractel - Toll Free SIP origination. Superior quality, unmatched reliability and awesome rates. Fractel is an FCC/SMS800 RESPORG and offers free number porting and vanity number search to qualifying accounts. ...
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Toll Free Termination Providers

January 11, 2013, 8:42 am
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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 such as HyperCube have a higher success in collection from LD carriers who support the RespOrgs CIC.

Page Contents

Without registration required



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.

FREE TOLL FREE GATEWAY: tollfree.alcazarnetworks.com:5060 (199.96.248.142)

Asterisk Dialplan Example

exten => _800NXXXXXX,1,Dial(SIP/${EXTEN}@tollfree.alcazarnetworks.com)
exten => _855NXXXXXX,1,Dial(SIP/${EXTEN}@tollfree.alcazarnetworks.com)
exten => _866NXXXXXX,1,Dial(SIP/${EXTEN}@tollfree.alcazarnetworks.com)
exten => _877NXXXXXX,1,Dial(SIP/${EXTEN}@tollfree.alcazarnetworks.com)
exten => _888NXXXXXX,1,Dial(SIP/${EXTEN}@tollfree.alcazarnetworks.com)


ArcTele Communications, Inc - VoIP Services
  • Free toll-free temination
  • No registration required
  • Send calls SIP 18XXXXXXXXX@tf.arctele.com:5060
  • Supports G729 ulaw
  • We send your Caller-ID.

Denetron Provides registrationless toll-free call termination over SIP and IAX.
  • CLID delivered as sent
  • Supported codecs: G711, G729a, GSM

Example Dialplan Configuration:
exten => _800NXXXXXX,1,Dial(SIP/sip.denetron.com/${EXTEN})
exten => _888NXXXXXX,1,Dial(SIP/sip.denetron.com/${EXTEN})
exten => _855NXXXXXX,1,Dial(SIP/sip.denetron.com/${EXTEN})
exten => _877NXXXXXX,1,Dial(SIP/sip.denetron.com/${EXTEN})
exten => _866NXXXXXX,1,Dial(SIP/sip.denetron.com/${EXTEN})


HyperCube is the leader in TFO or Toll Free Origination. They invented the process and support most providers in the USA that are able to receive a Network Connection Allowance (NCA)
HyperCube Website HyperCube or H3 is now a West Corporation. Toll Free Traffic destined to most US and CANADIAN providers is allowed.
The highest completion ratio in the industry and a 99.9% collection rate from RespOrgs/CICs. ...
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Elastix

January 11, 2013, 10:37 am
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Elastix is an appliance software that integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface. It also adds its own set of utilities and allows for the creation of third party modules to make it the best software package available for open source telephony.

The goals of Elastix are reliability, modularity and ease-of-use. These characteristics added to the strong reporting capabilities make it the best choice for implementing an Asterisk-based PBX.

External Links



Where to get support:

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VOIP Phones

January 11, 2013, 2:15 pm
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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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Analog Telephone Adapters

January 11, 2013, 2:22 pm
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Important notice to product manufactures, resellers and other people adding to this page



ALPHABETICAL ORDER PLEASE


Analogue Terminal Adapters (ATAs)

Please make sure you put entries into the correct category.
This Wiki is frequently visited by newbies who have never before done anything with telephony. Entries in the wrong category can lead to confusion. So, please refrain from marketing tricks getting your product listed as many times as possible. If some newbie buys your stuff only to find out it doesn't work the way it was expected and it won't work with his hardware or operating system it can only do harm. Adapters with an ethernet port go into category "FXS to Ethernet". Adapters with a USB connector go into category "FXS to USB". Further, in order to maintain fairness between all vendors and manufacturers do not use boldface or larger font to make your entry stand out and use only a single line for each product. If you want to provide a description longer than a single line, create a new Wiki page for your entry and point the entry to that Wiki. For more information on how to create a new Wiki page see How to add information to this Wiki. If you're just rebranding some else's ATA, please post somewhere else. Your cooperation is very much appreciated.

Analogue Phone to Ethernet gateways (FXS only - for devices with FXO ports see: VoIP Gateways)

The most common analogue terminal adapter is a device with at least one telephone jack (FXS port) used to connect a conventional telephone and an ethernet jack used to connect the adapter to the LAN. Using such an ATA it is possible to connect a conventional telephone to a remote VoIP server. The ATA communicates with the remote VoIP server using a VoIP protocol such as H.323, SIP, MGCP or IAX and encodes and decodes the voice signal using a voice codec such as ulaw, alaw, gsm, ILBC and others. Since ATAs communicate directly with a VoIP server, they do not require any software to be run on a personal computer, such as a softphone.

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Open Source VOIP Software

January 14, 2013, 1:37 pm
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Open Source VOIP applications, both clients and servers.

Open source means all source code is available!! Do not post any "free but not open" software here!

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SIP Proxies


  • JAIN-SIP Proxy
  • Mini-SIP-Proxy A very tiny perl POE based SIP proxy
  • MjServer cross-platform SIP proxy/registrar/redirect, written in java, based on MjSip stack
  • MySIPSwitch SIP Proxy server which allows using multiple SIP accounts with a single SIP login
  • NethidPro3.0.6 Opensource Sip Encryption Bridge: www.vonets.com
  • Net-SIP A Perl SIP framework that includes a stateless proxy
  • OpenJSIP Opensource distributed standalone SIP proxy, SIP registrar, SIP location service run by Java VM. ...
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STUN-bis

January 14, 2013, 1:38 pm
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Session Traversal Utilities for NAT (STUN)

The STUN protocol is currently has been rewritten with RFC 5389.

Changes to STUN Protocol


Originally, STUN (RFC 3489) was developed as a standalone solution for NAT traversal for several types of applications, including VoIP. However, practical experience found that the limitations of its usage in isolation made it impractical as a complete solution. There were too many NATs which didn't support hairpinning or which had address and port dependent mapping properties.

Consequently, STUN was revised:

  • STUN now is not the standalone solution for NAT traversal, but rather it defines basic tools and mechanism for performing the traversal. The broader and most comprehensive solution for NAT traversal currently is ICE, which makes use of the new STUN protocol.
  • now it is considered cutting-edge to NOT try to find out the type of NAT, and rather to discover the media path interactively using ICE. For SIP signaling, NAT traversal is being solved by the outbound draft.
  • several older STUN attributes have been deprecated since STUN no longer serves the purpose to detect NAT type, and new attributes and request/indication types are introduced. Examples of deprecated STUN attributes: RESPONSE-ADDRESS, CHANGE-REQUEST, SOURCE-ADDRESS, CHANGED-ADDRESS, REFLECTED-FROM, and MAGIC-COOKIE.
  • the new STUN also defines STUN usages, a mechanism to extend the STUN protocol. The STUN (draft) standard defines Binding Discovery and Binding usages. Other types of STUN usages include TURN.
  • the new STUN now provides means to authenticate STUN requests and responses by using short-term or long-term credential.


Current Standard


Successor to RFC 3489 is RFC 5389 http://tools.ietf.org/html/rfc5389


Implementations

  • PJNATH library from pjsip.org project is an Open Source NAT traversal library supporting ICE, STUN, and TURN.
  • STUN & TURN Server - is an open source STUN & TURN Server (and client library), for UNIX/Linux platforms.
  • Numb is a free STUN/TURN server.
  • libnice is a free GLib based ICE (draft 19), STUN (both RFCs) and TURN (draft 11) client implementation

SIP Clients


See also:


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STUN

January 14, 2013, 1:38 pm
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STUN (Simple Traversal of UDP through NATs (Network Address Translation)) is a protocol for assisting devices behind a NAT firewall or router with their packet routing. RFC 5389 redefines the term STUN as 'Session Traversal Utilities for NAT'.

Note: The STUN RFC states: This protocol is not a cure-all for the problems associated with NAT.


  • STUN enables a device to find out its public IP address and the type of NAT service its sitting behind.
  • STUN operates on TCP and UDP port 3478.
  • STUN is not widely supported by VOIP devices yet.
  • STUN may use DNS SRV records to find STUN servers attached to a domain. The service name is _stun._udp or _stun._tcp

Definitions (from the RFC)

  • STUN Client: A STUN client (also just referred to as a client) is an entity that generates STUN requests. A STUN client can execute on an end system, such as a user's PC, or can run in a network element, such as a conferencing server.
  • STUN Server: A STUN Server (also just referred to as a server) is an entity that receives STUN requests, and sends STUN responses. STUN servers are generally attached to the public Internet.

Various types of NAT (still according to the RFC)
  • Full Cone: A full cone NAT is one where all requests from the same internal IP address and port are mapped to the same external IP address and port. Furthermore, any external host can send a packet to the internal host, by sending a packet to the mapped external address.
  • Restricted Cone: A restricted cone NAT is one where all requests from the same internal IP address and port are mapped to the same external IP address and port. Unlike a full cone NAT, an external host (with IP address X) can send a packet to the internal host only if the internal host had previously sent a packet to IP address X.
  • Port Restricted Cone: A port restricted cone NAT is like a restricted cone NAT, but the restriction includes port numbers. Specifically, an external host can send a packet, with source IP address X and source port P, to the internal host only if the internal host had previously sent a packet to IP address X and port P.
  • Symmetric: A symmetric NAT is one where all requests from the same internal IP address and port, to a specific destination IP address and port, are mapped to the same external IP address and port. If the same host sends a packet with the same source address and port, but to a different destination, a different mapping is used. Furthermore, only the external host that receives a packet can send a UDP packet back to the internal host.



Closing words (also from the obsolete RFC 3489)

14.6 In Closing

The problems with STUN are not design flaws in STUN. The problems in STUN have to do with the lack of standardized behaviors and controls in NATs. The result of this lack of standardization has been a proliferation of devices whose behavior is highly unpredictable, extremely variable, and uncontrollable. STUN does the best it can in such a hostile environment. Ultimately, the solution is to make the environment less hostile, and to introduce controls and standardized behaviors into NAT. However, until such time as that happens, STUN provides a good short term solution given the terrible conditions under which it is forced to operate.





Standard documents

STUN RFC RFC 3489, now obsolete (Oct 2008)
STUN RFC RFC 5389 (Current as per October 2008)

Update to STUN protocol

STUN standard is currently has been rewritten with RFC 5389. ...
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Federated VoIP

January 14, 2013, 1:43 pm
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Background


Federated VoIP typically refers to:
  • Using ENUM to discover SIP and Jabber addresses associated with a particular phone number
  • Federated SIP, Federated XMPP/Jabber, or some combination of the two
  • Dynamically connecting from any one domain to any other domain without any pre-configured routing
  • Using the user's email address interchangeably as a VoIP/IM identifier (rather than forcing the user to have a Skype or IM address in addition to their normal email address)

Federated VoIP does not require any SIP trunks, intermediate VoIP providers, etc

Implementations of Federated VoIP


  • Unlike email, Federated VoIP (both SIP and Jabber) participants typically refuse non-TLS connections. In other words, using TLS is mandatory
  • Fortunately, both SIP and Jabber use the same type of certificates for Federated VoIP, so only one certificate is needed
  • Virtually all Jabber servers (for example, ejabberd) natively support the federation concept
  • SIP proxies typically support federation, but TLS support varies. repro and Kamailio are the optimal ones.
  • ENUM support is often an optional feature in SIP proxies, and it is not present at all in many Jabber clients
  • Software PBXes (e.g. FreeSWITCH, Asterisk) are NOT SIP proxies or Jabber servers: they are typically not suitable for federated VoIP on their own. They should be used as application servers in conjunction with a dedicated SIP proxy.

Technical overview


OpenTelecoms.org has a good technical overview with diagrams

SIP proxies suitable for Federated VoIP



Jabber servers suitable for Federated VoIP



Other aspects of a Federated VoIP deployment


It is highly desirable to provide all users within a Federated VoIP deployment access to a STUN/TURN server. The same STUN/TURN server will service both SIP and Jabber users:

  • reTurn from the reSIProcate project
  • TurnServer is another implementation of STUN/TURN
  • STUN & TURN Server - is an open source STUN & TURN Server (and client library), for UNIX/Linux platforms.

Provided that all phones/softphones support Internet Connectivity Establishment (ICE), STUN and TURN relay, the TURN server will ensure that users do not experience NAT problems. ...
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MyPBX U300

January 14, 2013, 10:52 pm
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IP PBX for SMBs

More Features, Less Cost!

MyPBX U300 boasts an embedded PRI (E1/T1/J1) port with up to 30 lines and 2 FXS ports in the one compact system, providing higher density trunking for offices using E1 PRI signaling. This system supports up to 100 users and 30 concurrent calls. Also it could be used as a gateway to legacy PBX systems in applications.

MyPBX_U300_01.gif
Product_function2.jpg


Specification:

Users: 100
Concurrent Calls: 30
Voicemail: Default 3000min

Interface:
One E1/T1/J1 ports (Support PRI,MFC R2,SS7), 2 FXS Ports
One RS232 Port
LAN: 1 (10/100Mbps)
WAN: 1 (10/100Mbps)
USB: 1 (USB2.0)
Audio In/Out: Supported

Size: 213x160x44 mm (1U half-width)
Weight: 1.2kg
Power Supply: AC 100-240V 50/60Hz 0.5A MAX

SIP Protocol: SIP(RFC3261), IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
Codec: G.711, GSM, SPEEX, G.722, G.726, ADPCM, G.729 A, H261,
H263,H263p, H264 ,MPEG4.
DTMF: Inband, RFC2833, SIP INFO
Echo Cancellation: Supported
Network: Static IP, PPPoE, DHCP, Firewall, VLAN, DDNS, QoS,
DHCP Server,OpenVPN
Multiple Languages Support:
Chinese, Dutch, English, French, German, Hebrew, Korean, Italian,
Polish,Portuguese, Romanian, Russian, Spanish, Thai, Turkish

Flow Chart

MyPBX_U300_FlowChart.gif


Features:

Attend Transfer
Blind Transfer
BLF Support
Blacklist
Callback
CDR (Call Detail Records)
Call Forward
Call Parking
Call Pickup
Call Routing
Call Transfer
Call Waiting
Caller ID
Conference
Define Office Time
DDNS
Dial by Name
DISA (Direct Inward System Access)
DIDs
Distinctive Ringtone
DND (Do Not Disturb)
FAX (T.38)
Firewall
Follow Me
IVR (Interactive Voice Response)
Intercom/Zone Intercom
Mobility Extension
Multi-language Prompt
Music On Hold
Music On Transfer
One touch record
OpenVPN
Phone Provisioning
PIN User ( PIN Code Control)
Paging/Zone Paging
PPPoE
Queue
Redundancy
Ring Group
Route by Caller ID
Skype Integration (Skype Connect)
SMS to Mail/Mail to SMS
Speed Dial
Three Way Calling
Voicemail
Voicemail to Email
Voicemail Forwarding
Web Based Control Panel
Spy functions (Normal Spy, Whisper Spy, Barge Spy)
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MyPBX U200

January 14, 2013, 10:54 pm
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IP-PBX for Your Business

All you need in one system!

MyPBX U200 is a workhorse designed for companies requiring up to 50 concurrent calls and 200 users. It supports PSTN, ISDN BRI lines, GSM/UMTS networks and VoIP. Also, it is equipped with an audio input port and an audio output port. MyPBX U200 is truly unrivalled for features and value.

Product_function2.jpg
MyPBX_U200_01.gif


Specification:

Users: 200
Concurrent Calls: 50
Voicemail: Default 3000min (upgradable)
Auto-Recording(Add-on): Support

Interface:
Up to 16 Analog Ports (FXO/FXS)
Up to 8 GSM Ports (Quad-Band GSM/GPRS850/900/1800/1900MHz)
Up to 8 UMTS Ports (UMTS 900/2100MHz or 850/2100MHZ or 850/1900MHZ)
Up to 8 BRI Ports
One RS232 Port
Flash: 512 MB Onboard Flash
RAM: 512 MB Onboard RAM
LAN: 1 (10/100Mbps)
WAN: 1 (10/100Mbps)
USB: 1 (USB2.0)
Audio In/Out: Supported

Size: 340x210x44 mm
Weight: 2.1kg
Power Supply: AC 100-240V 50/60Hz 1.5A MAX

SIP Protocol: SIP(RFC3261), IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
Codec: G.711, GSM, SPEEX, G.722, G.726, ADPCM, G.729 A, H261,
H263,H263p, H264 ,MPEG4.
DTMF: Inband, RFC2833, SIP INFO
Echo Cancellation: Supported
LED: Red for FXO/GSM/UMTS, Orange for BRI, Green for FXS
Network: Static IP, PPPoE, DHCP, Firewall, VLAN, DDNS, QoS,
DHCP Server,OpenVPN
Multiple Languages Support:
Chinese, Dutch, English, French, German, Hebrew, Korean, Italian,
Polish,Portuguese, Romanian, Russian, Spanish, Thai, Turkish

Flow Chart

MyPBX_U200_FlowChart.gif


Features:

Attend Transfer
Blind Transfer
BLF Support
Blacklist
Callback
CDR (Call Detail Records)
Call Forward
Call Parking
Call Recording
Call Pickup
Call Routing
Call Transfer
Call Waiting
Caller ID
Conference
Define Office Time
DDNS
Dial by Name
DISA (Direct Inward System Access)
DIDs
Distinctive Ringtone
DND (Do Not Disturb)
FAX (T.38)
Firewall
Follow Me
IVR (Interactive Voice Response)
Intercom/Zone Intercom
Mobility Extension
Multi-language Prompt
Music On Hold
Music On Transfer
One touch record
OpenVPN
Phone Provisioning
PIN User ( PIN Code Control)
Paging/Zone Paging
PPPoE
Queue
Redundancy
Ring Group
Route by Caller ID
Skype Integration (Skype Connect)
SMS to Mail/Mail to SMS
Speed Dial
Three Way Calling
Voicemail
Voicemail to Email
Voicemail Forwarding
Web Based Control Panel
Spy functions (Normal Spy, Whisper Spy, Barge Spy)
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VOIP Service Providers

January 15, 2013, 5:10 am
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For a list of VOIP to PSTN service providers, indexed by country, please see:


PLEASE DO NOT ADD NEW CATEGORIES HERE.

VOIP provider services, exchanges and other business deals belong under VOIP Service Providers B2B

Please keep your entry in ALPHABETICAL ORDER in relation to the other entries in your section.
If you add a new entry, including an 'added on dd/mmm/yy' would make it easier to notice.

Miscellaneous VOIP related services, including peer-to-peer services, are listed below.

Peer to Peer Service


  • SWISS VOIP Service Provider - HI Ring Someone - The worlds’ first virtually free VOIP platform to offer free and ultra-low cost international phone services with no contract and no monthly fee.Whether you are sitting down for a morning espresso in Hong Kong, a lunch date in Zurich or even dinner in Los Angeles, Hi Ring Someone offers free calling; anytime, anywhere.
  • VOIP Service Provider in India - Spectranet - leading Internet Service Provider Company in India offering VOIP Services in India, International Calling Cards, Prepaid VOIP Services, Business VOIP Services and all other affordable VOIP solutions in India
  • iVOIPE VOIP Service Provider in USA iVoipe is the leading next generation Internet Telephony Service Provider (ITSP) providing mobile and fixed services that dramatically reduce the cost of roaming and international calling. ...
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VoIP Innovations

January 15, 2013, 5:16 am
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VoIP Innovations - Wholesale VoIP Carrier
VOIP INNOVATIONS
"Your Premier Wholesale VoIP Carrier"


WEBSITE:

Learn more today, http://www.VoIPInnovations.com


ABOUT US:

VoIP Innovations is a wholesale VoIP Services provider, servicing ITSP's and resellers, through our industry leading back office, Titanium III. If you are looking for wholesale DIDs, origination, SIP termination, or E911 services, VoIP Innovations is your carrier. Our VoIP wholesale services allows reseller access to all the major VoIP carriers with unprecedented aggregation, control, and automation. Let us negotiate and manage your telecommunication network. VoIP Innovations won't lock you into a long term contract or ask for an unreasonable minimum.

VoIP Innovations believes in using technology to automate daily business and operational processes. We are constantly developing new tools to deliver more automation and transparency to our customers. Our propriety back office, Titanium III, enables service providers to quickly provision wholesale VoIP services in real-time and includes a full suite of tools to automate service delivery. Titanium III allows management of all your services through an easy to use web portal.


ADVANTAGES:

•Aggregation: We create strategic relationships with top telecommunications carriers, leveraging our high-volume traffic to negotiate aggressive rates. What does that mean for you? Nothing less than the best possible wholesale pricing in the industry.

•Coverage: Ever-expanding tiers of service. A footprint over 8,500 rate centers strong. More than 500,000 DIDs available in our number warehouse for instant provisioning. It all adds up to one of the nation's largest VoIP networks, servicing the majority of the United States and Canada.

•Automation: We are constantly developing new tools to make service delivery faster and easier. Our BackOffice portal gives you the ability to manage your account online, anytime. Everything from ordering DIDs to local number porting to accessing billing and support-available to you 24x7.


COVERAGE:

Worldwide - US and over 60 countries. Learn more, About VoIP Innovations


NEWS :

•2011-06-26 - http://www.voipinnovations.com - VoIP Innovations announces the release of Titanium III. A stellar platform gets even better!
•2011-05-25 - http://www.voipinnovations.com - VoIP Innovations announces the expansion of their number warehouse from 120,000 DIDs in stock to over 500,000.
•2011-04-14 - http://www.voipinnovations.com - VoIP Innovations integrates a new LCR SW providing lower termination rates and better quality.
•2011-03-17 - http://www.voipinnovations.com - VoIP Innovations will be attending the ITExpo West in Austin. Visit us at booth #914
•2010-09-10 - http://www.voipinnovations.com - VoIP Innovations announces the release of Titanium their premier wholesale VoIP platform.


RESOURCES:

If you'd like to learn more about what VoIP technology can do for your business, just visit our VoIP Resource Library, http://resourcelibrary.voipinnovations.com/.

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Asterisk consultants USA

January 15, 2013, 1:35 pm
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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!

Page Contents



ALABAMA


Asteria Solutions Group

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Kamailio Consultants

January 15, 2013, 11:28 pm
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Add your entry here (Alphabetical order by company):
This page is growing large. Please don't post logos!!

Page Contents

Asipto

  • Web site: http://www.asipto.com
  • Daniel-Constantin Mierla, co-founder OpenSER/Kamailio
  • Elena-Ramona Modroiu, co-founder OpenSER/Kamailio
  • OpenSER/Kamailio services
  • OpenSER/Kamailio solutions
    • Internet Communication/Telephony Platform (Voice, Video, Instant Messaging and Presence)
    • SIP Prepaid System
    • Standalone Nat Traversal Box
    • LCR, Load Balancer and Traffic Dispatcher System (Scaling PSTN Gateways, Asterisk, FreeSWITCH, etc...)

CCIE Task Force

CCIE Task Force Consulting Services provides Cisco Certified Internetwork Experts of all disciplines,
CCIE Routing & Switching, CCIE Voice, CCIE Security; each of which are certified professionals in their field.


Dr. SIP (Zürich, Switzerland)

Expert for SIP, ENUM, Linux, MySQL, OpenSER/Kamailio, and Asterisk

  • 3rd Level Support for SIP Issues / Troubleshooting
  • Installation and maintainance of SIP (SER/OpenSER/Kamailio, Asterisk, and other) systems
  • High availability architectures
  • Linux and MySQL
  • NAT traversal
  • Integration of heterogeneous VoIP systems
  • ENUM integration
  • Service contracts
  • Strategic product direction and IETF Standardization
  • Reviewing and development of architectures and designs
  • Education

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VOIP Service Providers B2B

January 16, 2013, 2:43 am
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Here is a list of VOIP Service Providers focusing on Business-To-Business services. This includes VoIP origination and VoIP termination, plans aimed at call centers, IVR providers and generic Asterisk users. See also:

Services which require the use of locked ATA devices should not be listed on this page. Nor should services which do not permit simultaneous calls — most services here support at least 4 simultaneous incoming calls. Please list only services which support Asterisk connections, via SIP or IAX2, to the PSTN.

Page Contents

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VOIP Service Providers Residential

January 16, 2013, 3:06 am
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DID Service Providers

January 16, 2013, 3:08 am
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A Direct Inward Dialing service provider delivers a telephone number over VoIP protocol including SIP, IAX2 or h323. This DID Phone Number will receive calls using a sip soft phone, hard phone, or an IP PBX. The charges are charged per month, and per min, and per channel based. You do not require a hardware card in case you have a DID service provider to receive call, ie a PRI Card or a Analog Card. The call travels to you all the way over the Internet

Cheapest DID Providers


Algeria

  • Algerian DID numbers Currently only national Algerian numbers.. Also 68 other countries available. Free forwarding to Voip/SIP, IAX,H323 Google Talk, etc many providers preconfigured. Forwarding to PSTN (landlines and mobiles) from 2cent/min, including mobiles. Mobiles Europe 6cent/min.

Argentina

  • Phone2call Virtual Numbers DID are available in 60+ countries including Argentina. Free forwarding to Voip/SIP (some preconfigured providers), IAX, H323, Google Talk, Virtual PBX (yes, free Hosted PBX). Also, very cheap call forwarding to PSTN (regular landlines and mobiles). No contracts. Pay per month, Instantaneous Activation, Many payment methods. Try not only our Virtual Numbers DID but also our free Hosted PBX which has many features (useful for example for Small and Medium Business). Know why we are known for our VOIP leadership.
  • Argentina DID Numbers€ 4.99/month| DomesticNumbers offers Argentina virtual phone numbers from 19 different cities in Argentina, including Buenos Aires. Website also in Spanish.

Australia


Austria

  • Sipgate.at Free personal 0720- national number in Austria.
  • TeleCallMart Unlimited Incoming Phone Numbers, Voip calls, SIP Phone, Auto Attendant, DTMF. No monthly fees. Low prices!


Bahrain

  • Phone2call Virtual Numbers DID are available in 60+ countries including Bahrain. Free forwarding to Voip/SIP (some preconfigured providers), IAX, H323, Google Talk, Virtual PBX (yes, free Hosted PBX). Also, very cheap call forwarding to PSTN (regular landlines and mobiles). No contracts. Pay per month, Instantaneous Activation, Many payment methods. Try not only our Virtual Numbers DID but also our free Hosted PBX which has many features (useful for example for Small and Medium Business). Know why we are known for our VOIP leadership.
  • Bahraini DID numbers Currently only regional Manama numbers. Also 68 other countries available. Free forwarding to Voip/SIP, IAX,H323 Google Talk, etc many providers preconfigured. Forwarding to PSTN (landlines and mobiles) from 2cent/min, including mobiles. Mobiles Europe 6cent/min.



Brazil

  • Brazil Free DIDs Free Brazil DIDs. Limit 5 DIDs per IP. Free SIP Forwarding Only. Click on Free DID section on our site to request. ...
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Sip Trunking Providers

January 16, 2013, 7:24 am
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This page is a list of SIP trunking providers around the world. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.

Country specific pages:

VoIPInvite IncWholesale SIP termination service for dialer traffic. Call center using predictive dialer are welcome. 1000 of calls per second. no LRN, just npa nxx billing available. NPA NXX start from $0.0015. Canada, USA, international dialer termination available.

1comms VoIP provider for UK Businesses. Sip trunk services, UK DDI's and Asterisk Support for business users and resellers. White Label fully itemised per second billing.

1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.

Alcazar Networks - Wholesale Services Over 3,100 DID rate centers. Per minute pricing as low as $0.0005/minute. Per channel pricing as low as $2.00/channel. DIDs as low as $0.10/each. A-Z termination. Over 1,200,000 numbers in DID inventory.

Amivox free your phone - Lower your communication cost VoIP provider for both consumers and businesses. Offer's free SIP account. Prepaid and very good rates for network termination with premium quality ( Amivox-Out) . Support for iPhone, Android and Blackberry. Shared balance for multiple users. Calling Amivox to Amivox is free - Sign up for free and try out the service.

Anveo offers phone numbers from over 48 countries with instant activation. Anveo's Voice 2.0 Communication and Collaboration Suite with powerful Visual Call Flow technology allows you to visually configure call handling and call termination options for your phone number. Anveo provides FREE SIP trunking and it is one of many termination options available.

BellVoz offers International and Domestic Long Distance Services with VoIP technology, helping business and consumers to reduce monthly telephony expenses.

Best VoIP USA BestVoIPUSA.com offers SIP trunking to private and commercial operators of Asterisk PBX switches. BestVoIPUSA.com also provides all of the well known features of SIP even if you are connecting directly to SIP compliant devices, handsets or servers.

Box Internet Services offers SIP trunking to private and commercial operators of Asterisk PBX switches. Boxis.net also provides all of the well known features of SIP even if you are connecting directly to SIP compliant devices or servers.

Brisnorth Communications Australia Brisnorth.com.au provides SIP trunks, VoIP and SIP Server hardware to Businesses Australia-wide. Carrier-Grade reliable SIP/VoIP services at very cost-effective rates. We can work with your current hardware/phones or upgrade you. We have Plans to suit all budgets and sizes of Business. ...
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