Benifits Of Voip News
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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, Wireless internet phone jackson tn service
configurations, standards, tips & tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony. However, tivo vonage call
Wiki is primarily for information, not for advertising.
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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 - 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 full series from Yeastar.
2010-03-10 - Apstel Visual Dialplan for Asterisk - Run Through with Screenshots.
2010-03-10 - Phone System Comparison Chart for Spring 2010 Released by CompareBusinessProducts.com - over 90 phone systems compared
2010-03-10 - VoIPon Interviews Rhino Equipment Corp. ... SIEMENS SIPART 6DR2004
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 networks 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 skype to your mobile phone
AudioCodes - FXS & FXO
Axtan
AZACALL200 - 2 port FXS, 1 port FXO, 1 Lan , 1 WAN. SIP ATA with router and FXO functionality
Boscom Boscom Claro range 2, 4 or 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 vonage commercial song please 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 Agent
http://www.AsteriskAgent.com
World Wide Asterisk Support
support@AsteriskAgent.com
800-763-2908
Specializing in Asterisk based solutions.
Asterisk Support & Staffing
Live Website 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 proyecto VOIP ...
VOIP GSM Gateways
What's a VoIP GSM Gateway?
A 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 messenger pc to phone
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 pc to phone iraq 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 internet phone india - SIP/H323 VoIP GSM Gateway / IP PBX
2N VoiceBlue Lite - VoIP GSM Gateway Vonage Reviews
Large GSM Gateway for termination - 2N StarGate ISDN PRI GSM IP gateway with VoIP card
2N Traffic really free internet phone
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 gateway 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 Test
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 => 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)
; #=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 => longdistance
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. ...
Asterisk AGI phpivr
You can get phpivr from here
1. Copy folder ivr into astagidir (/var/lib/asterisk/agi-bin/)
$ cp -rv ./ivr /var/lib/asterisk/agi-bin/
2. Create symlink on IVR.php
$ ln -s /var/lib/asterisk/agi-bin/ivr/IVR.php /var/lib/asterisk/agi-bin/IVR
3. Copy configuration file cheap pc to phone india and phpivr_extensions.conf into /etc/asterisk/
$ cp phpivr.conf phpivr_extensions.conf /etc/asterisk/
4. Edit configuration file. By default IVR will run menu named "common".
Parameter `options' - required minimum to play "welcome" message.
voip agent reseller
/>5. For use Default settings copy folder ./sounds/ivr ( from demosounds.tar.bz2) into asterisk sounds dir
$ cp ./sounds/ivr /var/lib/asterisk/sounds
6. Add these lines into /etc/asterisk/extensions.conf
; IVR vonage isp />#include "phpivr_extensions.conf"
then from asterisk CLI type following
asterisk*CLI> dialplan reload
7. Change folders owner to asterisk user
$ sudo chown astreisk.asterisk /var/lib/astreisk/agi-bin/ivr /var/lib/asterisk/sounds/ivr
8. Call to SIP/7777 number and test IVR. By default 0, 2 and 1 menus (dtmf inputs) are available.
1 - jumps to another menu, 0 - exit from menu, 2 - playing additional info. Other dtmf - exit.
See Also
phpivr configuration example
phpivr project at SF.net
vonage theme
- Trac
Page visited: {HITCOUNTER} times
phpivr configuration example
Suppose that we need to do interactive voice responses, which briefly say in which company received a call and then redirect the call to the Sales department (1001) if you press 1, Corporate clients dep. (1002) by pressing 2. Well, as usual - to connect with the Secretary (1000) if you press 0 or wait while cheap pc to phone iraq
message ends. Also we provide the case when from the terminal will introduce dtmf not covered by the menu – play message “to hear the menu again” then run it again.
To achieve this simple task, we need:
sound file with greeting - welcome.gsm
voip investors
sound file with describing the available menu (it can be combined with a greeting) - free internet phone calls to india
a sound file with a request to hear the menu again and make your choice - please-make-your-choice.gsm
Put the recorded files into the folder /var/lib/asterisk/sounds/ivr and set them corresponding rights
in phpivr.conf write
{
"common" : {
"name" : "Main menu of my company IVR"
,"options" : "say=ivr/welcome|say=ivr/main-menu,prompt,loop=1|transfer=1000"
,"inputs" : {
"0" : "transfer=1000"
,"1" : "transfer=1001"
,"2" : "transfer=1002"
}
,"inputnotfound_act" : "say=ivr/please-make-your-choice|menu=common"
}
}
From here it should be noted that while playing a welcome message, pressing any key will cause it to completion, and immediately will playing main-menu. Also you can omit the option loop=1 because the count of playback by default is 1, but It written by me for a better understanding of the written material.
That's all the settings. Enjoy.
See Also
phpivr AGI installation guide
trixbox
vonage router problems
and running in about 15 minutes
This page has been viewed {HITCOUNTER} times since being created on {CREATIONTIME}
Page ContentsFeatures
Official Resources
News
IP PHONE
Service Providers
Resources
Tutorials
System Installers
Argentina
Australia
Canada
China
Colombia
France
Germany
India
Iran
Israel
New Zealand
Pakistan
PUERTO RICO
Russia
Singapore
Slovak republik
Turkey
UK
Ukraine
USA
See also:
trixbox is an open telephony platform utilizing the best of the open source telephony tools into one easy-to-install package. Based on an enhanced LAAMP (Linux, Apache, Asterisk, mySQL, PHP) the trixbox dashboard provides easy to use broadvoice promo code
interfaces to setup, manage, maintain, and support an complete IP PBX system.
Succeeding Asterisk@Home 2.8, trixbox has been loosely described as 'Asterisk@Home 3.0'. It offers improved stability and the promise of an upgrade process that doesn't require you to wipe voip providers chicago
entire install and start over.
Polycom support for trixbox now includes Paging and Intercom, Buddy Lists, and BLF functionality.
Features
trixbox 2.8 contains the following:
CentOS 5.3
Asterisk 1.6
DAHDI
mySQL
Apache
PHP
PBXconfig 5.5
VoIP Setup Wizards
Admin status screen
Network configuration tool
Telephone provisioning for Linksys, Polycom, Snom, Grandstream, Cisco, and Aastra
much much more
Official Resources
trixbox support options - paid support available directly from Fonality/trixbox
internet telephony voip
Asterisk sound files requests
Asterisk sound files requests
Digium's The Voice web site lets you pay for custom IVR prompt recordings by Allison.
You may also go directly to Allison's web site
This web page is a place to put requests for sound files to be recorded by Allison via The Voice. These requests are batched up in order to reach the 1 hour minimum that she processes. Costs recovered through Paypal.
Westany provides custom voice prompt recordings in 18 languages. as vonage song
as full custom languages sms pc to phone calls
request.
Voice Studio offer fast Australian prompt recording. No minimum and pricing starts at $15. Use 'VOCUSTOM' as a coupon code to save 5%. Voice Studio Web Site. Packs for Asterisk, 3CX and pbxnsip also available.
Voice Vector Media (http://www.VoiceVector.com) offers complete replacement voice-packs for Asterisk with over 1,500 prompts each. Inexpensive custom recording services are also available. As of December 19, 2006, the company's website is offering custom recordings at half price.
Richmedium, through a partnership with a professional recording studio, offers custom sounds. Average delivery is 2 business days. Check the website for samples.
Elianna lets you pay for custom sounds by Elianna in Spanish (Argentina or neutral), check the webpage Elianna recordings
Obviously, writing your requests here is no guarantee that they will get done. The more generic the sound files, the more likely they are to be included.
Back Trax Inc Custom Promts Recording Great Quality at Great Prices! Email Back Trax 301-948-9671
If you need sound files in german - have a look at www.pforzheim.de/asterisk they give them to the community for free! (:biggrin:)
Asterisk Sound Files for Patches
The requests through December 2004 have have been completed, and are available for download at http://www.geekthing.com/~robf/asterisk/. The recordings there are free to use, copy, and redistribute freely. They are now in asterisk-sounds CVS.
Add new requests here:
Languages
... in English.
... in French.
... in Spanish.
... in German.
... in Italian.
... in Mandarin.
... in Cantonese.
... in Japanese.
List of other common languages available here: http://en.wikipedia. ... vonage commercial song
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