login | register
Sat 17 of May, 2008 [14:22 UTC]

voip-info.org

Search with Google
Search this site with Google. Results may not include recent changes.
 
Google Ads
Shoutbox
  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • 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.
Server Stats
  • Execution time: 0.39s
  • Memory usage: 2.19MB
  • Database queries: 33
  • GZIP: Disabled
  • Server load: 0.58

Touchstone

http://www.touchstone-inc.com/

VoIP verification solutions for SIP and H.323 including call generation, analysis and QoS monitoring.

Net Observer remote monitoring and diagnostic suite announced!

Download your Free VoIP Analyzer HERE!

Professional grade VoIP testing solutions starting at just $495!

From Website:

Touch·stone (tu­ch'stone)

1. An excellent example that is used to test the excellence or genuineness of others.
2. A fundamental or quintessential part or feature.
3. A standard by which something is judged.

Touchstone Technologies has been developing essential Voice and Video call generation and network monitoring and analysis solutions since 1989. Specializing in delivering exceptional value in state-of-the-art test and measurement products, Touchstone is completely committed to our customers' success. But we don't expect you to take our word for this, we invite you to put us to the test! Contact us and try our products at your convenience, in your environment. See for yourself how Touchstone's products can provide all of your essential testing solutions for a fraction of the price of other options. Want more details? Schedule one of our private, informal, web demonstrations at your convenience!

WinSIP SIP Call Generator

WinSIP is a high-performance SIP call generator providing advanced signaling complete with audio and video streams. Fully configurable and ideal for connectivity, load, stress, and feature testing, WinSIP is unrivaled in its combination of power, portability, flexibility and affordability.

WinSIP is a completely self-contained, 100% software-based product. WinSIP can support 3,000+ simultaneous G.711 calls and up to 5,000 G.729 calls with perfect media streams from a single PC. For stress and load testing, WinSIP can support 45,000+ signaling calls at an astonishing rate of 5,000,000+ busy hour call completions (that's right, 5 million call completions per hour; not call attempts!). WinSIP is also completely scalable, offering an ultra high-density solution at a fraction of the cost of hardware-based systems.

WinSIP is more than just a call generator, it can be used as:

A Call Initiator, scripted to run up to 45,000 calls simultaneously.
A Call Terminator, scripted to answer up to 45,000 calls.
An Unattended Answerer (unscripted, 45,000 calls).
A Lightweight Proxy Server.
A Registrar load and stress testing tool.
A Media path through SBC verification tool.

See for yourself why WinSIP has become the SIP traffic emulator of choice for companies such as Ericsson, Sonus Networks, Siemens, AT&T, Sprint, MCI and many, many more.


WinEyeQ Voice and Video over IP Network Monitor and Analyzer

WinEyeQ is a state-of-the art SIP and H.323 network monitoring and analysis solution for voice and video over IP. With its Data Scope presentations and call-centric architecture, WinEyeQ provides 10 unique viewpoints of your network. From a high-level network overview to endpoint equipment and activity, no other offering gives you quite the breadth and depth of detail that WinEyeQ does. WinEyeQ provides the essential features necessary to monitor, analyze, diagnose, record, replay, "trap", alert and alarm signaling and media events. WinEyeQ also offers detailed reports, network and call metrics, call quality assessments, media QoS analysis, and hundreds of relevant VoIP and protocol metrics. WinEyeQ's unique monitoring/analysis duality will save your engineers countless man-hours diagnosing and troubleshooting problems as well as providing an ever-vigilant early warning system for voice, video, or triple-play network.




How To Debug and Troubleshoot VOIP
Created by jht2, Last modification by Tzafrir Cohen on Wed 16 of Jan, 2008 [16:59 UTC]

Comments Filter

Alphabetical Order

by Nezer on Saturday 15 of October, 2005 [16:21:54 UTC]
Mark,

When you're adding your site to the ordered lists found around the wiki please be sure to follow the guidelines and post in alphabetical order. I removed your entry on VOIP Sites because it was out of sequence. Feel free to add it back in it's proper place.

Thanks.

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