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...ype Outage - VoIP on mobile devices - or perhaps something more personal..... Come one, come all. Bring your story. Connect details at http://vuc.me Michael Graves -- Michael Graves mgraves<at>mstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgraves at mstvp.onsip.com skype mjgraves Twitter mjgraves

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

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

...g 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 years so changing these now will merits the voip consideration. Thanks, Michael -- Michael Graves mgraves<at>mstvp.com o713-861-4005 c713-201-1262 sip:mjgraves at pixelpower.onsip.com skype mjgraves fwd 54245

...ccount. 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 pixelpower.onsip.com skype mjgraves fwd 54245

...; 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 54245 >> >> >> >> > > > >The 430 has a larger footprint and feels to be a slightly heftier and >more business-like phone. I have very similar sound quality and results >with either phone. T...

.... Junction told me that they have a bounty on offer for this if someone's interested in doing the work. Would the availability of the Firefox code make it easier to do an ActiveX implementation? Any takers? Michael -- Michael Graves mgraves<at>mstvp.com o713-861-4005 c713-201-1262 sip:mjgraves at pixelpower.onsip.com skype mjgraves fwd 54245

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

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

On Tue, 4 Jan 2011, Andy Graybeal wrote: >> The Polycom 321 has not been EOL'd and supports VLAN. It is, however, >> lacking a 2nd ethernet port if you were to go that route. >> >> -M >> > Thanks for the response Mark. I see the 331 has two ports and the same > features as the 321. > > I'm wondering what phone would be best being used as an

I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine. What could be the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 echo the sentiment that you should just run Asterisk on some small hardware....in an appliance like fashion. In fact, just yesterday I posted an overview of hardware suitable for DIY appliances. I've used many of the platforms mentioned. http://www.mjgraves.com/2010/07/08/d-i-y-asterisk-appliances-a-question-of-scale/ Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgraves at mstvp.onsip.com skype mjgraves > -------- Original Message -------- > Subject: Re: [asterisk-users] Pbx_f?r_Windows?_-_Email_found_ > in_subjec...

...bx_f?r_Windows?_-_Email_f ound_ in_subject I echo the sentiment that you should just run Asterisk on some small hardware....in an appliance like fashion. In fact, just yesterday I posted an overview of hardware suitable for DIY appliances. I've used many of the platforms mentioned. http://www.mjgraves.com/2010/07/08/d-i-y-asterisk-appliances-a-question-of-scale/ Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgraves at mstvp.onsip.com skype mjgraves > -------- Original Message -------- > Subject: Re: [asterisk-users] Pbx_f?r_Windows?_-_Email_found_ > in_subjec...

...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. mgraves at mstvp.com o713-861-4005 c713-201-1262 skype mjgraves fwd 54245

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

...rpath defends their implementation as being according to the RFC. This begs the question of what does Asterisk do with G.722? I've yet to try v1.6 so I open the question to the group. 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 fwd 54245

.... The call will happen Friday Nov 7 at 12 noon EST. To find out more about how to join this call please visit: http://blog.mgraves.org/ or http://voipusersconference.org/ning/ Michael Graves -- Michael Graves mgraves<at>mstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mjgraves at pixelpower.onsip.com skype mjgraves fwd 54245

Anybody care to muse on Wi-SIP vs. SIP-DECT? My limited research indicates that none of the WiSip phones will ever be able to match the performance of DECT phones. Maybe I'm wrong but a Wi-SIP phone seems like a DIESEL sports car. There is nothing wrong with the technology, but it seems like a shoe-horned fit into the requirements of a wireless endpoint. DECT uses a wireless radio layer