similar to: [VOIP-Users-Conference] Re: Opinions please: Polycom IP 430 vs 330?

I need to add a few phones to an existing installation. They have a dozen IP430 at the moment. Does anyone feel that there are advantages to the IP330? Cost is not the major consideration as long as they're in the same range. (under $175) Michael -- Michael Graves mgraves<at>mstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:mjgraves at pixelpower.onsip.com skype mjgraves fwd

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.

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

Does anyone on-list have any experience with this system? I'd like to speak with someone who's tried it out. Thanks, Michael -- Michael Graves mgraves<at>mstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:mjgraves at pixelpower.onsip.com skype mjgraves fwd 54245

Which flavor of G.722 has been implemented in Asterisk? And starting with what release version? Thanks, Michael -- Michael Graves mgraves<at>mstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mjgraves at pixelpower.onsip.com skype mjgraves 54245 at fwd.pulver.com

I'd like to survey those on-list who actually use wifi SIP handsets. What type of wifi access point do you use? Are you happy with it? I presently use some older Linksys WAP54G APs. I'd like to replace these but in doing so I'd like to be moving in a VOIP friendly direction. I've yet to find a handset that I'd buy in quantity, but my last round of access points lasted >4

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

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 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 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

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

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

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,

How difficult is it to change the firmware load on a Polycom phone from MGCP to SIP? I have a number of 500/600 phones and see some used phones being offered with MGCP installed. I have the SIP firmware but have never had to migrate between content loads. Any expected gotcha's? Michael Graves -- Michael Graves mgraves at pixelpower.com Sr. Product Specialist

There came up an interesting question on last weeks VoIP Users Conf call. http://vuc.me. An internet broadcaster wanted to be able to take a number of "phone calls" and bring them into a traditional audio mixer where each call was a separate balanced audio source to the mixer. Let's say that he might need 8 simultaneous calls. Is there a way to set this up using a single Asterisk

Hi All, My appologies for the off-topic post, but I thought it would still be of interest to this list. If you've been reading our site at http://vuc.me you've no doubt seen that we have the video call with LifeSize scheduled for Feb 4th. What? You didn't know? Here's t scoop: http://www.voipusersconference.org/2011/hd-video-conferencing-with-lifes ize/ We have a number of

There simply aren't that many local Asterisk Users Groups. This is one of the reasons that the VoIP Users Conference (http://www.VoIPUsersConference.org) has been picking up members. Using the Talkshoe conference bridge is certainly accessible, and the price is right (free!) I listened in on the streams from 25C3 over the Christmas break, and fout it really enjoyable. It occurs to me that

Can anyone here have experience using G.722 on the Grandstream GXP-1200? It's supposed to support the codec, but I wonder if the handset does it justice? The older BT-200 also supported the codec, but the handset was not good enough. You could only hear the improved call quality using a headset. Michael Graves mgraves <at> mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgraves at

Hi all, Many thanks for your replies! Will tell my friend and see what he will be interested in. Many thanks! Christian -----Ursprungligt meddelande----- Fr?n: mgraves at mstvp.com Till: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Skickat: 10-07-09 15:29 ?mne: Re: [asterisk-users] Pbx_f?r_Windows?_-_Email_f ound_ in_subject I

Does IAXy2 have any internal diaplan? What I need to do is have a phone that when picked up automatically dials out in a hotline fashion. I know that I can setup Sipura devices to do this, but I'd rather use IAX2. Alternatively, do any of the cheapish SIP phone support this sort of internal dialplan functionality? Thanks, Michael -- Michael Graves