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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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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 Eric Chamberlain on Wed 23 of May, 2007 [23:18 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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