similar to: wifi

I'm in process of transitioning a number of offices to a hosted virtual pbx from Junction Networks. It's a combination of OpenSER and Asterisk. They have a nice click-to-call extension for Firefox, but I need the equivalent for IE so that it can work with our CRM system. Junction told me that they have a bounty on offer for this if someone's interested in doing the work. Would the

Doing a little research before Friday's Voip Users Conference call with Dan Behringer. Are any of the newer Polycom wideband codecs implemented in v1.6? Specifically, G.722.1 or G.722.2? Thanks, Michael Graves mgraves <at> mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgraves at mstvp.onsip.com skype mjgraves

I'd like to add a device to my Asterisk server to leverage my cellular account. Does anyone on-list have experience with hardware gateways vs using cah_bluetooth and an old cell phone? I'm considering something like http://www.mobigater.com/index.php?p=5 Thanks, Michael -- Michael Graves mgraves<at>mstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:mjgraves at

On this weeks VUC call we will collectively be our own guests. That is, we'd like to know what was the big issue that impacted YOU in 2010? All opinions welcome. Here are a few things to get you thinking in advance: - Apple's Antenna-gate - Asterisk 1.8 Launches - Amazon EC2 as a DOS platform - Cisco launched UMI video conference device - More HDVoice capable phones - Skype Outage - VoIP

Does anyone online have an opinion on these? I've used 500/510/6001/601 models before. Need to know if these apparently lesser models can be provisioned in the same way. Are end uers happy with them? Thanks, Michael -- Michael Graves mgraves at pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

Hi all, We have put together a firmware for a range of inexpensive routers. It has been configured to provide optimum VoIP performance. We have internally tested it for number of months and it looks very good. You should be able to run it easily with 20+ phones on local network ( we still did not hit the upper limit ) assuming that you have bandwidth. Your VoIP will get prioritized over other

This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards,

Has anyone here ever had the occasion to setup a system that would dynamically alter it's codec preferences based on trafffic? That is, presuming that the system is on a limited bandwidth connection is would start to prefer a compressed codec as the call volume increased? Perhaps shifting from G.711 to G.729? Michael -- Michael Graves mgraves<at>mstvp.com http://blog.mgraves.org

Hi All, A couple of weeks ago I noticed Askozia PBX, which is a new embedded Asterisk & OS distro at http://askozia.com/pbx. This caught my attention for two reasons; it uses v1.4 of Asterisk, and it uses the m0n0wall development framework to build on FreeBSD with a PHP based GUI. I've used m0n0wall for years, and FreeNAS also, which shares the same OS/GUI framework. I booted the latest

Hi All, So I've been exploring the use of G.722 encoded wideband audio recently. I have three different SIP devices that allow this: Eyebeam, IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine together. Calls pass between them in what the Polycom notes as "HD" mode and the audio quality is certainly very good. However, things are not so easy with Eyebeam and the

Hi, I would be curious to play with an Android phone with Wifi-only capability. My plan is to install Bria on it and see if it could be used within a couple of WiFi access points, as a high-end wireless phone. Which handset would you recommend ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP range in the U.S. I'm particularly interested in the Gigaset S685 IP. Since it's DECT 6.0, and there's an English (UK) version, I'm thinking it should work just fine, after dealing with the walwart issue (and maybe caller ID signalling). Anyone imported one from the UK and using it in the US? for how

I see a variety of DECT 6 phones available CHEAP at costco. Is there a way to convert these to SIP? I recall someone talking about a Siemens devices that works with all DECT phones, making them SIP (I think).... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081205/fdd14af9/attachment.htm

I will have a small shop with ~4 phones using an HP server with Asterisk on it, it has two NICS and so I planned on plugging one into the cable modem, and the other into the switch. I was going to let this box perform NAT for the company but I am concerned about QOS for the VOIP portion. Anyone got a similar setup and care to share what they successfully implemented? Thanks! jlc

Hi, Try http://gigaset.siemens.com/shc/1,1935,hq_en_0_11729_rArNrNrNrN,00.html you will have good selections there. Regards, Vidura Senadeera ================================ To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <200711252201.lAPM1NUP005677 at mail871.megamailservers.com> Content-Type:

Hi there, in case anyone is interested, I've just taken ownership of a small home network (3 handsets) of the brand new Siemens DECT PSTN/VOIP phone. It works great with Asterisk. Here's my overview and review so far... http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ Cheers Al -- The way out is open! http://www.theopensourcerer.com

Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't figure out why the volume is so low. How can I adjust the volume control on Asterisk? It's at max on the handset phones. Thanks! Hin

I recently got a Pulver Innovations WiSIP wireless SIP phone to determine if we want to use them in our organization. Since the WiSIP phone arrived last week, I have had nothing but headaches. I do think I now have the problem narrowed down. I have spent a bulk of my time trying to get the WiSIP to work with a couple of Linksys WAP11 Version 2.2. I have met with no success in getting the WiSIP

While I myself am not a MagicJack user, I'm curious as to whether anyone here has managed to heist their MagicJack account's sip credentials, and use them to terminate calls using asterisk. Apparently it's as simple as sniffing the SIP credentials. If so, said person would enjoy unlimited termination for $20 year while retaining the flexibility of setting their CallerID to a

Michael Grave just posted a question about surround conferences. http://www.facebook.com/notes.php?id=564633430#/note.php?note_id=5009726 3908&id=564633430&index=0 I didn't see it posted on the ast-list, what do you think? Does something like this have potential? I'd love to listen in on one of these calls to see how it actually sounds if someone builds a trial