At the risk of continuing off-topic conversation... Oh the M9 has it's own issues, don't you worry (not to mention it's _way_ more expensive than the Gigaset range)!
I've been testing the A510 today and I've decided I like it more than the A580. The software (via the web interface) looks more polished and overall thus far it has performed fairly well (with no annoying 'new message' statement!). I haven't tried multiple SIP registrations with dual handsets yet, but I intend to. You guys realise these are standardised DECT handsets right? Ie that's where the 6-handset limitation comes from, and you should theoretically be able to register other manufacturers' handsets too. I'm about to try the C610IP model that Kai-Uwe mentioned (over the next few days). I think I'll like even more than the A580 since it has a higher resolution colour display. And even then it's still miles cheaper than the M9! Don't get me wrong, I do really like Snom phones too, although the M9 hasn't been entirely smooth sailing. I love the 320 desk phone (at the cheaper end of the scale, it has a great feel to it). Isn't it great that they all interact with Asterisk so well (there you go, I'm on-topic again ;-). Pete On 12/12/2012, at 4:12 PM, Mitul Limbani <[email protected]> wrote: > Mebbe you guys should try snom m9 dect ip phone, i have been using it since > over 3 years now without any of these issues. > > Mitul > > On Dec 12, 2012 4:25 AM, "Kai-Uwe Jensen" <[email protected]> wrote: > Using a Gigaset C610IP here, and am very happy with the features. The base > station can handle two concurrent SIP calls, and another internal one at > that. It does it with a single SIP registration to each server. You can setup > multiple servers if you want to and define dial patterns/plans that determine > which server gets used. After some playing around with it, I'm now using my > setup connected to a single asterisk only. (Let asterisk make call routing > decisions based on cost, using an AGI) > > Call transfer is working fine, the handsets have a Flash/R key to accomplish > this. Using the Flash lets you start a second call, and once answered you can > easily conference the second party in (softkey right on the screen), or > transfer the call to the other party (via menu, then transfer). Using this > capability, someone on a call can easily confer with another party, and > bridge them into the call. AFAIK it is not possible for someone to join an > existing call easily. You'd have to implement that in asterisk's dialplan, > not on the Gigaset phone. > > My understanding is that the C610IP has a few more features than the 510. I > might've also read somewhere that the 510 is obsolete. Can't find that link > right now, but search mgraves.org (use the Gigaset tag to get some initial > results). > > > On Tue, Dec 11, 2012 at 2:37 PM, Roy Abshire <[email protected]> wrote: > That is true about the A580. > > I don't like the interface much to check messages. > > Besides that every time I go to dial a number...it always uses the first > digit pressed to go into phone mode..so I have to press the first digit > twice... > > I would test other phones but it's for home and I can't fork over $$ to try > them all out.... > > I have tested some Nokia cell phones, the N97, N900, and E71 and the E71 and > N900 worked well. I didn't like the N97. > > > Co-op Vacation Rentals > > www.coopvr.com > > 15218 Summit Ave > Suite #300-354 > Fontana, CA 92336 > Phone/Fax (855) 760-COOP (2667) > > On 12/11/2012 12:52 PM, Pete Mundy wrote: >> One thing I dislike about the A580H is that the handset always says 'You >> have new messages' if I've missed a call. It wouldn't bug me if it said >> 'missed call' but it tells me I have new messages and even lights up a red >> LED under a button with a picture of an envelope on it. >> >> I'm about to test an A510IP and an A610IP to compare against the A580. >> Fingers crossed neither of them has that issue, because the Gigaset phone is >> a pretty good phone other than that, and the difficulty doing a (blind) >> transfer, as referred to by the OP. >> >> Pete >> >> >> On 12/12/2012, at 8:57 AM, Roy Abshire >> <[email protected]> >> wrote: >> >> >>> I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. >>> Never gave me problems. The call Quality is excellent! >>> I only have 1 handset connected to the Base but I want more. I bought a >>> Linksys WIP330 as a 2nd phone to try out and that works just as good >>> without a base unit. >>> >>> The A580 Base supports up to 6 handsets. >>> >>> I have 6 Incoming VOIP Numbers using separate SIP accounts pointed to 1 >>> Handset but you can point each SIP to separate handsets. >>> >>> The call goes to the first phone that picks up. When on a call, picking up >>> another phone makes a separate call and does not conference. I don't use >>> conference yet but I know you have to put the call on hold or something. >>> >>> The thing I don't like about the A580 and might be the same on all of them >>> is that you can only specify 1 Sip Account for making outgoing calls. In >>> other words, all 6 phones would use the same caller id out, but I wanted to >>> be able to choose that because I have a business number and number for each >>> person in our household. In order to use a different Caller ID (SIP >>> Account) for making outgoing calls I added a extension to my Dial Plan and >>> before making outgoing calls I press *1-6 before the number. >>> >>> I'm going to try adding more handsets that are compatible. I want the >>> SL78H but they are so expensive for just home everyday use. >>> >>> Make sure you check the compatibility page here before buying handsets. >>> >>> >>> http://gigaset.com/us/en/cms/PageCustomerServicesCompatibility.html >>> >>> >>> >>> Co-op Vacation Rentals >>> >>> www.coopvr.com >>> >>> 15218 Summit Ave >>> Suite #300-354 >>> Fontana, CA 92336 >>> Phone/Fax (855) 760-COOP (2667) >>> >>> On 12/11/2012 11:32 AM, sean darcy wrote: >>> >>>> Siemens A510IP >>>> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by >> http://www.api-digital.com >> -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> >> http://www.asterisk.org/hello >> >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
smime.p7s
Description: S/MIME cryptographic signature
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
