login | register
Fri 09 of May, 2008 [19:46 UTC]

voip-info.org

Navigation
Today's Deals
Recently Visited Pages
Search with Google
Search this site with Google. Results may not include recent changes.
 
Google Ads
Last Changes
Shoutbox
  • 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.
  • Christopher Faust, Wed 07 of May, 2008 [15:28 UTC]: When I try to startx I ge input not supported. Though before installing asterisk I had no video issue to start the GUI
  • Christopher Faust, Wed 07 of May, 2008 [15:26 UTC]: Hi Nick, I got centos 5.1 and asterisk up But now I cannot start startx I have set the depth from 24 to 16 for the video i810 driver for the i845 on my netvista machine but I cannot start GNOME. Please advise
  • Nick Barnes, Wed 07 of May, 2008 [10:01 UTC]: Howard - You'll need to provide a lot more information if you really want help.
  • Nick Barnes, Wed 07 of May, 2008 [10:00 UTC]: Christopher - Search the Wiki and you'll find a page I wrote detailing exactly what you have to do for Asterisk 1.4 + CentOS 5.1.
Server Stats
  • Execution time: 0.43s
  • Memory usage: 2.39MB
  • Database queries: 43
  • GZIP: Disabled
  • Server load: 1.12

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.

Your contributions are welcome, but please read the How to add information to this wiki page and the Posting Guidelines before you post.

NEWS



News Resources


Getting Started


Connecting Phones to VOIP


Connecting VOIP to the PSTN and Cellular Networks


PBX and Servers - VOIP PBX and Servers

Please post new/other servers here, because they will be removed.
  • Asterisk: Open Source PBX
  • Bayonne: Open source PBX
  • FreeSwitch: Open source cross platform (*nix, Win, OSX) multiprotocol (SIP, IAX, Googletalk, Jabber, XMPP, WOOMERA/H.323, PRI/T1/E1) softswitch.
  • CallWeaver: Vendor independent, cross-platform open source IP-PBX, originally derived from Asterisk.
  • OpenSER: Flexible and powerful open source GPL SIP ( RFC3261 ) server with integrated TLS
  • Sip Express Router: High-performance, configurable, free SIP ( RFC3261 ) server
  • sipX Solution Summary: SIPfoundry - The SIP PBX for Linux (L-GPL)
  • YATE - Open Source Linux/Windows GPL Telephony Server and Client (has support for SIP, H.323, IAX2, E1/T1, voicemail), H.323 - SIP translator.
  • more...

VOIP Misc.


Protocols - the language of VOIP


Markup Languages

  • Basic call routing and rules for UA's or VOIP serversCPL
  • IVR Presentation and dialog management: VoiceXML, CallXML
  • Call control / conferencing / call routing: CCXML
  • IVR / Speech recognition definition: SRGS
  • IVR / Speech synthesis definition: SSML
  • IVR / prompting / recording / conferencing / DTMF / Voice: CallXML

Traditional Telephone Network


VOIP Events and Conferences


VOIP Websites: Other VOIP websites on the Internet


Suggestions and Questions


Hit counter

This page has been viewed 9286575 times since being created on Sat 01 of Oct, 2005 16:47 UTC

RSS Feeds

  • Image Page Changes
  • Image Comments
Created by System, Last modification by Jack on Fri 09 of May, 2008 [13:15 UTC]

Comments

Comments Filter

LOOKING FOR LATIN AMERICA WHOLESALE ROUTES

by Luis Sanz on Monday 28 of April, 2008 [15:38:08 UTC]
Hello ,
I'm Luis Director of sales for tpt telecom, we are base out of the u.s California. Im looking to purchase new routes or sale. A-Z Terminations as wholesale, specially looking for Latin America but well consider any others.
Please Provide me with a list of your rates or what you looking for and contac info so we could do some business.

Thanks
Luis Sanz
TPT-TELECOM
Director of sales
310-848-7029
sanzinvestments@gmail.com

FRAUD ALLERT!!!

by Voipswitch on Tuesday 15 of April, 2008 [15:44:39 UTC]
Please note SOLUTIONS4VOIP has STOLEN voipswitch software and is trying to illegally sell it, Voipswitch software belongs to Voiceserve Inc a US company that is listed on the US stock market, voiceserve has filled all its papers with the government department of the SEC in the US in regards to its ownership of Voipswitch, please note solutions4voip is based in Pakistan and has NO contact number on its website, beware they steal minutes as well!

Please beware SOLUTIONS4VOIP also trade under APNAVOIP.

Legal proceedings have now started against them and anybody else buying hacked versions of Voipswitch.

for more information regarding voipswitch software please visit www.voipswitch.com

by Rupa Schomaker on Tuesday 15 of April, 2008 [10:25:41 UTC]

Fight SPAMMERS, boycot them !

by Mats Karlsson on Saturday 05 of April, 2008 [17:18:11 UTC]
Fight SPAMMERS! If they cant read and understand the posting guidelines !

"Advertising or shameless promotion in the comments section of any page on this site is considered SPAM. and will be reported.
If you have any questions or comments please email support@voip-info.org. Thank you."

How could they be trusted as a company ?

B O Y C O T T H E M ! ! ! !



kamran = apnavoip = Unstrustworthy company

by Mats Karlsson on Thursday 27 of March, 2008 [12:38:08 UTC]
They are a NOT a trustworthy company, they continuously SPAM this web site and has been told not to do so...

Dont do business with them !

kamran, this is not an advertising space!

by Mats Karlsson on Thursday 27 of March, 2008 [12:34:31 UTC]

by linkx on Wednesday 19 of March, 2008 [06:03:36 UTC]

Joseph Arsenault

by Joseph Arsenault on Wednesday 05 of March, 2008 [03:09:19 UTC]
Just wanted to introduce myself, I'm Joseph Arsenault and I love asterisk. I'm currently using astlinux on a CF card on an old Toshiba SG10 which is a celeron 366mhz with 256mb and a 64Mb CF card.

by linkx on Wednesday 27 of February, 2008 [11:39:06 UTC]
Roy, not to my knowledge, but check the post date.

HTML Supported SIP Phone

by mezukhan on Wednesday 27 of February, 2008 [10:14:44 UTC]
Anybody knows any sip phone that has HTML browser...

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:
Terms of Service Privacy Policy
© 2003-2008 VOIP-Info.org LLC

Powered by bitweaver