Vonage In The News Phone Service Corp News
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New Software Releases
Page ContentsArchive
March 2010
February 2010
January 2010
This page is to inform on various VoIP related software releases.
Archive
2007
2008
2009
Your contributions are welcome, but please read the How to add information to this wiki page and the Posting Guidelines before you post.
March 2010
2010-03-12 - REMWAVE OS X Communicator as open source - REMWAVE releases source code for SIP softphone for Mac OS X
2010-03-01 - Orgasmatron 5.1 for Asterisk - Free U.S./Canada phone calls plus 30 free VoIP apps (15-minute install)
February 2010
2010-02-24 - SafiServer and SafiWorkshop Version 1.3 Beta - Sipadan
to the opensource graphical IVR designer and server engine (for Asterisk)
2010-02-22 - Taridium ipbx Taridium releases ipbx eXpress free 5 user edition
2010-02-20 - HostingVOIP release new vippie for voice & SMS VIPPIE for Windows, Symbian, Andriod & Windows Mobile.
2010-02-20 - OfficeSIP Softphone 1.0 is released. Free voice & video softphone for Windows.
2010-02-15 - DTH Software, Inc releases version 5.0 of its vonage wav advertisement
Billing System. Release Notes.
2010-02-12 - UnitePBX. ...
VOIP Service Providers Business Europe
This is a list of VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:
VOIP Service Providers Residential Single line residential/business plans go here.
VOIP Service Providers B2B Bulk origination/termination goes here.
RIP VOIP VOIP provider cemeteryâ™
Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations migrates to switch site mobility voip bts rel services
consumer protection laws of that country.
Marketing is NOT ALLOWED on this page. Please describe services in neutral language and normal fonts. Don't bother listing prices--unless you really plan to return and edit them as things change. If you want to add your company, please read the Posting Guidelines for Promoting Products and Services. When you add your entry to this page, please make sure your entry is in alphabetical order in relationship to other vendors listed in the same section. Failure to follow these guidelines will result in deletion!
Please include relevant information like SIP or IAX handoff, how outgoing CLID is set, whether a given account may originate multiple calls at once, etc.
If you like this page, voip radio dispatch
link to it, so Google and other search engines will consider it more important.
Users: Please feel free to REMOVE any listing that does not meet the stated goals of this page. ... vonage 'minimum bandwidth'
VOIP Service Providers Residential
Service Providers Residential
Africa
Logic Ring Very competitive International VoIP Reseller programs for agents and private label resellers.
IPtransit Residential and SME Sip, H323, SIP URI, Inbound, LoCall Plans, PC and Mobile Phone clients, well suited to VSAT, SCPC and contended bandwidth clients, 723.1, &29, 711 and true T38, Calling cards and GSM Gateways, BYOD welcome, test accounts.
Corelynx Inc Hosted Enterprise Telephony for SME and Residential, Full Suite of Call Center Solution - Hosted and Onsite Model, Offers DID and 8XX numbers to more than 50 countries of the world. Capable of offering VOIP based IP PBX Solution on MPLS and VPN for countries where VOIP ports are blocked, IPLC and Colocation Services, A-Z wholesale and retail termination. Also offers any form of customization work on Asterisk or SER platforms.
INDIGICOM VoIP Solutions Provider. Business services, end-users, satellite VoIP.
MyKanKan VoIP Service Provider. Business solutions for call-centers, hotels, callshop .... Voip and Asterisk consulting. Features include : Voicemail, Caller ID w/Name, Call Waiting, Call Forwarding, Caller ID Block, Free In-Network Calling, Web Based Call Logs. Optional Services include: SoftPhone Access, Virtual Phone Number.
http://www.powerbillbox.com African VoIP carrier and billing provider. African PSTN calling and peering, national & international rates in H323, SIP, IAX.
Takalam Takalam offers VoIP solution starting from $10 to small and medium business in all African and Arab countries like Saudi Arabia, UAE, Jordan, Egypt, Bahrain, Kuwait, Iraq, Syria, Palestine, Kenya, Nigeria, Rwanda, etc...
TuT Telecom - ABSOLUTELY FREE Peer-to-peer SIP calls vis softphones or SIP hardware. Local DIDs also available to over 7000 cities. Asterisk Supported. PC2Phone services and voip services. WiFi & GSM Voip Phones and services available.
South Africa
Asterisk Directory South Africa Comprehensive Listing of Asterisk Companies and VoIP providers in South Africa
Logic Ring Very competitive International VoIP Reseller programs for agents and private label resellers.
FutureFone South African VOIP provider. SIP based account. ... x-pro sip configuration
voip-info.org
Welcome to the VOIP Wiki - a reference guide to all things VOIP
This Wiki covers everything related to VOIP, software, hardware, service providers, reviews, configurations, standards, tips & tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony. However, the Wiki is primarily for information, not for advertising.
Notice: Editing pages to eliminate SPAM and inappropriate content is appreciated, when there are disagreements about what is appropriate content please contact: support@voip-info.org.
If you are removing content from pages, please leave a polite comment explaining the reason for the changes. H.323 Analyzer comments, user names, or content will result in the removal of the users editing privileges.
Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelines before you post.
NEWS
2010-03-12 - Asterisk Open Source Conference coming up...
2010-03-12 - Cisco claims CRS-3 IP Routing will revolutionize the internet...
2010-03-12 - VoIPon Brings the Best of CeBIT to You VoIPon Interviews Sangoma, AVM, Pika, Patton, snom, Grandstream
2010-03-11 - Announcing the 24-hour VoIPathon VoIP chat live to celebrate the three years of VoIP Users Conference
2010-03-11 - DTH Software, Inc releases new Express edition of its billing system for start-ups. 1-24 concurrent calls.
2010-03-10 - Auto-provision function now starts to appear on small embedded IP PBX: MyPBX closed cell sip construction series from Yeastar.
2010-03-10 - Apstel Visual Dialplan for Asterisk - Run Through with Screenshots.
2010-03-10 -
Asterisk RealTime PostgreSQL
So, rather than have this stuff all over the place, I thought it would be easier if I just made a page that would handle all things RealTime and PostgreSQL.
While PostgreSQL can still be used Asterisk cdr pgsql, it can no longer be used with the vonage phone service review system. Thus, if you are using the old pgsql for your db driven voicemail, when you upgrade, you will have to use the unixODBC code. As such, I figured it was just as easy to use the business and voip
subsystem for the CDR. I haven't done any preformance testing yet, but my guess is that the over head is not enough to justify having two different DataAccess layers. Anyhow, I choose to use one, so here is how I got RealTime and the CDR subsystem to use unixODBC.
These are the tables that I used to make my Asterisk RealTime work. I also am including a modified cdr table. I am planning on using the system with an online replicator (http://www.commandprompt.com), which means that all tables must have a primary key. As such, I added the primary key to the cdr table create statement. Lastly, I am put in some internet phone calls using MCI number
(for us anyway :P) defaults. I also changed the tables to reflect some of the "size" constraints that we run into. We tend to have very long appdata sections for our extensions since we use the app_sql to do things in the database from music from vonage commercial
the dialplan.
WARNING: Asterisk versions prior to 1.4.15 suffer a vulnerability in res_config_pgsql. If you want Postgres realtime, update immediately to 1.4.15!
NOTE: We are broadband phones companies PostgreSQL 8. ...
VoIP Gateways
If you want to add your company's products, please read the Posting Guidelines for Promoting Products and Services
Media Gateways
Media gateways, also commonly referred to as VoIP gateways are devices which bridge conventional telephone free internet phone
and equipment to VoIP telephone networks. A typical media gateway has at least one conventional telephone port and at least one ethernet port.
Analog FXO gateways
in alphabetical order
1Telecom Ltd - FXO/FXS VoIP GSM Gateways
2N Telekomunikace - FXO GSM Gateways
Aastra Aastra Venture FXO Gateway
Abilis Abilis the all-in-one VoIP gateway with ISDN backup
AirTouch - FXO/FXS Skype VoIP Gateways
Allwin Tech SIP/H.323 dual protocols, 2/4/8 FXO/FXS ports ,NAT, Router, register up to 4 servers simultaneously
Anketechnology - FXO gateway--VoicePixie-211 www.anketechnology.com
Atcom - FXO gateway for skype Au-600forward broadband phone service to your mobile phone
AudioCodes - FXS & FXO
Axtan
AZACALL200 - 2 port FXS, 1 port FXO, 1 Lan , 1 WAN. SIP ATA with router and internet telephony voip functionality
Boscom Boscom Claro range 2, 4 vonage stock
8 port FXO
Camrivox - FXS & FXO
Cisco - FXS/FXO
http://ciscosystems.wordpress.com
http://comstore.us
http://ciscomemory.net
D-Link DVG-3004S: 4 FXO port trunk gateway for SIP
[http://www.pikatechnologies.com/english/view.asp?x=607
Asterisk Paid Support
Important notice to posters entering new companies to this page:
If you want to add your company, migrates to switch site mobility voip bts rel services
read the Posting Guidelines for Promoting Products and Services
Please, preserve alphabetical order.
Page ContentsAsterisk Agent
AsteriskExpert
Asteriskneeds
Asterisk-Schweiz.ch (Switzerland)
AsteriskService.com
Asteriskware
ATY Consulting
Bitnetix Incorporated
credativ Ltd.
.e4 Technologies
Emergen Consulting
Enterux
Foehn Ltd
Kibrit.net Freelancer Network & Security Group
Linux Technology Group
LOD Communications
NetHawk (pvt) Ltd.
PC Depot Inc (Technology for non-profits and you)
Ponesupport
Support.Asteriskguru.com
Troop Software
US VoIP Systems
Voicemeup
People Tech systems pvt limited
Asterisk vonage song
http://www.AsteriskAgent.com
World Wide Asterisk Support
support@AsteriskAgent.com
800-763-2908
Specializing in Asterisk based solutions.
Asterisk Support & Staffing
Live clip art voice over IP
Support Available
AsteriskExpert
http://www.AsteriskExpert.co.uk
Digium Certified Asterisk Professional (dCAp)
Asterisk, Queuemetrics, Vicidial, Call Centres, references available.
Installation, Support, Maintenance, One Off Incidents or service contracts. ...
VOIP GSM Gateways
What's a VoIP GSM voip books
VoIP GSM Gateway enables direct routing between IP, digital, analog and GSM networks. With these devices (fixed cellular terminals) companies can significantly reduce the money they spend on telephony, gp-especially the money they spend on calls from IP to GSM. The core idea behind cost saving with VoIP GSM Gateways is Least Cost Routing (LCR).
Through least cost routing the gateways select the most cost-effective telephone connection. They check the number which is dialed as well as rate information which is stored in an internal routing table. Because several SIM cards and GSM modules are integrated within the VOIP GSM Gateway it is able to make relatively cheaper GSM to GSM calls instead of expensive IP to GSM calls.
Who offers VoIP GSM Gateways?
Page Contents2N TELEKOMUNIKACE
2N VoiceBlue Enterprise - SIP/H323 VoIP GSM Gateway / IP PBX
2N VoiceBlue Lite - VoIP GSM Gateway
Large GSM Gateway for termination - 2N StarGate ISDN PRI GSM IP gateway with VoIP card
2N Traffic Control Manager
Acecom Teles
Teles IGate GSM Gateway E1 Channel Bank 32 SIMs (Only for Sale in AsiaPacific)
Abilis the all-in-one VOIP gateway
Cost Saving Effect:
USB interface:
Dial Through:
Other features:
Cyber-Telecom.net.
ELGATO Communications
VOIP GSM gateway ELGATO 32 (8-32 GSM-channels)
VOIP GSM vonage in the news phone service corp
ELGATO 16 (4-16 GSM-channels)
SIM-Server ELGATO
SIM-Bank ELGATO
Hypermedia Systems Ltd. ...
Asterisk tips ivr menu
Interactive voice response menus
Implementing a simple 'push-1, push-2' menu structure
The key to creating this menu is to create an Extension (defined as 205 below) to record your menu prompts. This will put the sound file in /tmp/asterisk-recording.gsm. You'll have to move that file each time its created to /var/lib/asterisk/sounds and rename it to something pertinent to your design so it can be called from the dial-plan. Notice the line under [mainmenu] exten => s,5,Background(sai-welcome). The
sai-welcome is one of those .gsm sound files. The rest of the dial-plan just defines what happens when each option is pushed. If you want to be able to have regular users update the voice prompts, see asterisk tips phrase recording menu.
extensions.conf
[mainmenu]
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo test
exten => s,5,Background(sai-welcome)
exten => s,6,Background(sai-choose)
; Tech Support
exten => 1,1,AGI(dima-test.agi)
exten => 1,2,SetGlobalVar(ACCOUNTCODE=${callerid})
exten => 1,3,SetVar(testcallerid=${callerid})
exten => 1,4,Background(sai-reptech-welcome)
exten => 1,5,Queue(rep-tech)
; Leave Voicemail
exten => 2,1,VoicemailMain()
exten => 2,2,Hangup
; Echo sip home kits exten => 3,1,Playback(demo-echotest)
exten => 3,2,Echo
exten => 3,3,Playback(demo-echodone)
exten => 3,4,Goto(mainmenu,s,6)
; EAGI Test
exten => 4,1,Answer()
exten => 4,2,Wait(1)
exten Sipix SC-1300 digital camera manual 4,3,AGI(sai-repid.agi)
exten => 4,4,Wait(1)
exten => 4,5,Hangup
; Play Music-on-Hold
exten => 5,1,MusicOnHold(default)
exten => 5,2,Goto(mainmenu,s,6)
4 wire E&M multiplexer VOIP
; #=hangup
exten => #,1,Playback(sai-thanks)
exten => #,2,Hangup
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
[default]
include => mainmenu
include => local
include => chevy vonage
include => joe-iax
include => npi-iax
; Record voice file to /tmp directory
exten => 205,1,Wait(2) ; Call 205 to Record new Sound Files
exten => 205,2,Record(/tmp/asterisk-recording:gsm) ; Press # to stop recording
exten => 205,3,Wait(2)
exten => 205,4,Playback(/tmp/asterisk-recording) ; Listen to your voice
exten => 205,5,wait(2)
exten => 205,6,Hangup
Example menu with timeout and invalid option. Works with Asterisk 1.6
exten => s,1,Set(NUMINVALID=0)
exten => s,2,Set(NUMTIMEOUTS=0)
exten => s,3,Background(thank-you-for-calling)
exten => s,4,Set(TIMEOUT(digit)=5)
exten => s,5,Set(TIMEOUT(response)=10)
exten => s,6,WaitExten(5)
exten => t,1,Set(NUMTIMEOUTS=$[${NUMTIMEOUTS}+1]})
exten => t,2,Gotoif($["${NUMTIMEOUTS}" < "3"]?s,3)
exten => t,3,Background(vm-goodbye)
exten => t,4,Hangup()
exten => i,1,Set(NUMINVALID=$[${NUMINVALID}+1]})
exten => i,2,Gotoif($["${NUMINVALID}" < "4"]?:10)
exten => i,3,Background(invalid)
exten => i,4,Goto(s,3)
exten => i,10,Playback(vm-goodbye)
exten => i,11,Hangup()
Implementing a high-density without wearing out your keyboard
Now consider an information delivery IVR, such as a bus schedule. ...
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