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Sat 17 of May, 2008 [07:14 UTC]

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  • 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.
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AudioCodes

http://www.audiocodes.com/

AudioCodes Ltd. enables the new voice infrastructure by providing innovative, reliable and cost-effective Voice over Packet technology and Voice Network products to OEMs, network equipment providers and system integrators. AudioCodes provides its customers and partners with a diverse range of flexible, comprehensive media gateway, server and processing technologies, based on VoIPerfect™ - AudioCodes’ underlying, best-of-breed, core media gateway architecture.

Products & Solutions

AudioCodes' product line includes Enabling Technology Products such as Voice over Packet chip processors, VoIP PCI and cPCI Telephony boards & media gateway modules, Core Network Products for the wireline, wireless, broadband access and enhanced voice services applications and CPE & Access Gateways, comprised of AudioCodes analog and digital media gateways. These products are integral of the most advanced and reliable Voice over IP and Voice over ATM platforms on the market, and have been implemented successfully by leading telecommunications and networking manufacturers worldwide.

Voice over Packet Processors
Media Gateway Modules
VoIP & Telephony Communication Boards
CPE Analog Media Gateways
CPE Digital Media Gateways
High Density Media Gateways: Mediant Product Family for the converged, wireline, wireless and broadband markets
Media Servers
EMS (Element Management System)
Session Border Controllers (SBC) & Security Gateways

Where to buy

Support Service

  • Binary Systems, Inc. - Full support for all AudioCodes products from the original team that created and administered the AudioCodes Certified Partner Program. Binary Systems has trained the master distributors, OEMs, and carriers in SIP, H.323, MGCP, MEGACO, TPNCP, and Netrake Session Border Controller IMS/UMA Security solutions. Service Contracts and Installation Services available. Email sales@binary-systems.com or call 972-238-9146.

  • POST CTI - In addition to offering competitive pricing on our wide range of hardware and software products, e.g AudioCodes, we at POSTcti also take pride in our ability to offer a range of standard and bespoke Professional Services. From initial architecture design consultation, through installation configuration and solution testing to post-sales support available 24 hours a day, 365 days a year. Call us on +44(0)870 1266633

Created by vmdigioia, Last modification by Michael Diks on Thu 15 of May, 2008 [10:37 UTC]

Comments Filter

Working MP-118 and Trixbox

by Jeremy Mann on Tuesday 19 of June, 2007 [19:21:21 UTC]
Since I'd prefer not to repost here, I created a success story about trixbox and the MP-118.
<a href="http://www.trixbox.org/forums/trixbox-forums/share-your-trixbox-success-stories/mp-118-and-trixbox-integration-success">Trixbox Forum Post</a>

by stephen on Wednesday 18 of April, 2007 [18:48:33 UTC]
ok, now that i did all that, where did my caller id go. as an aside how can the caller id be transfered to another extension instead of the extension number showing uP? (in asterisk sorry alittle off topic)

by stephen on Wednesday 18 of April, 2007 [14:23:35 UTC]
>The idea being that the MP is only a gateway, if you have an IP side Proxy...ie ASTERISK, then you should datafill the Endpoint numbers with the Asterisk >AutoAttendant number.

I am using asterisk. i only use pstn for now. is turning off autodial better? thanks than my "fix"?

by stephen on Wednesday 18 of April, 2007 [14:16:39 UTC]
Yada thanks.
I solved my dial out by changing Protocol Definition>Gerneral Selection>Channel Select to something else. No it dials not just then endpoint number. I will try you third statement, how is that diffent. Will try to hunt group and report back...

Daling IP--TEL on a FXO

by Yada on Wednesday 18 of April, 2007 [04:15:14 UTC]
Really simple.

1. Under Hunt Groups. Define a hunt group (1) with channel select of ascending
2. Under Endpoint Phone numbers : Assign the hunt group fields to 1
3. Under Routing tables: Route any source and any destination number to Hunt Group 1.

There are variations to this using the different channel select methods, routing tables and Manipulation Tables, with multiple hunt groups, but the above should get you on your way.

PS. Make sure the FXO Settings are set for 1 stage dial. Unless you want PBX dialtone to play and force you to dial again. Some older PBX's might need you to push out time before dialing from 1000-2000 ms.

Re:

by Yada on Wednesday 18 of April, 2007 [04:10:35 UTC]
Stephen,
read my 3rd statement from the 28th of March.

If you want dial tone when the audiocodes picks up, then disable Auto Dial under the Endpoint settings. then you can dial what you want... up to max digits.

The idea being that the MP is only a gateway, if you have an IP side Proxy...ie ASTERISK, then you should datafill the Endpoint numbers with the Asterisk AutoAttendant number.

Re:

by Yada on Wednesday 18 of April, 2007 [04:08:58 UTC]
Stephen,
read my 3rd statement from the 28th of March.

If you want dial tone when the audiocodes picks up, then disable Auto Dial under the Endpoint settings. then you can dial what you want... up to max digits.

The idea being that the MP is only a gateway, if you have an IP side Proxy...ie ASTERISK, then you should datafill the Endpoint numbers with the Asterisk AutoAttendant number.

by stephen on Sunday 15 of April, 2007 [18:40:48 UTC]
I

by stephen on Sunday 15 of April, 2007 [17:56:23 UTC]
I have it almost working it will only call a phone number if that number is in the ENDPOINT SETTINGS field. whats up with that. what option needs changed. thanks....

by stephen on Friday 06 of April, 2007 [15:14:21 UTC]
bump :> I still have a little hair left but cannot dial out to the mp114 any pointers would be great. in works ok but out, phone will dial and not connect. get party unav message from trixbox

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