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>From atacomm.... ________________________________ From: Jessee J Holmes [mailto:jholmes@atacomm.com] Sent: Thursday, May 12, 2005 2:24 PM To: Wiley Siler Subject: Re: Got a date yet? No specific release date as of yet; but, we're hoping to have a physical date soon. So far planned release is either...

Has anyone used the Polycom HDvoice phone yet? I am curious if it uses a different codec. Does it actually sound any better?

Hello, I just bought a Snom-320 from ATAComm. I plugged it into my LAN, registered it with *, etc. All my other SIP gear is Sipura and works fine, both on the LAN and over the Internet. The new Snom seems like it can't process the audio from the handset mic. A steady tone is garbled, even on the LAN. I've contacted ATAComm, Sn...

I am having the same issue as below. Has this issue been solved or does anyone know an answer? This error recently began and we have multiple phones out of commission. PLEASE HELP!! http://lists.digium.com/pipermail/asterisk-users/2006-August/162841.html How did you find out about 468*??? It's sure as poop not documented in the Polycom Admin Guide anywhere. -----Original Message-----

Froogle found me one supplier for the SPA-841, not sure I trust them though. Does this phone even exist yet? Does anyone have any experience with it? Does anyone know a vendor other than Atacomm/voipsupply?

A client of mine has a Snom 320. Usually when he comes in each morning, it is asking him for a password. A power cycle brings it back to normal operation. How do I troubleshoot this further? --Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070221/0e4d3ab8/attachment.htm

We must have had the magic version of 1.6.x then, because we increased our buddy watch limit from 8 to 48 in that version. -----Original Message----- From: Jessee J Holmes [mailto:jholmes@atacomm.com] Sent: Wednesday, October 11, 2006 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list Limit was increased in firmware 2.0.1. NOTE: a new Polycom Administrator's guide is now also av...

Polycom 601 with Sip 2.01 Anyone using Sip 2.01? I have upgraded my phones and now presence no longer functions. Buddy list shows all phones online but status does not change when someone is on a call. Also blf does not function. I am using trixbox, 1.67 was working fine on the same box. Any ideas?

Can someone please help me with a problem that I seem to have with this Polycom 601 phone. It will not see my TFTP server and keeps saying "Could not contact boot server, using existing configuration". I have Linksys phones that use the TFTP server without any problems but this Polycom will not see or use it. Please Help. -------------- next part -------------- An HTML attachment

...rg> wrote: > What I don't understand is why people MUST use the 2.0.x firmware. I believe it's because the limit of how many can be monitored at once, someone correct me if I'm wrong. > > Jessee J Holmes wrote: > > A few of our technical support staff here at Atacomm are currently > > working on this issue with Polycom, Digium and one of our customer's > > (who posted in here earlier this week). > > > > There are some major differences from the 1.x.x firmware and the 2.01 > > firmware. Obviously, many on here, voip-info.or...

Does anyone have an actual delivery date on the new Polycom HD IP650's? I'm getting sick of not having a backlit screen and thinking of upgrading. Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061018/9937184f/attachment.htm

It's noisy while talking. Any idea? Thanks in advance. Jason ____________________________________________________________________________________ Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates (http://voice.yahoo.com)

Hi, Where should I go to get the Polycom`s latest official (non-beta) version? I am registered on the Polycom customer website but that doesn't seem accessible. Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061102/284b67a7/attachment.htm

Hi all, Has anyone had any experience getting Polycom phones repaired? The screen on one of our IP600s got smashed, and I'm wondering if it's worth the effort to get it repaired, or if it'd just be cheaper to buy a new phone. Thanks, -- James Andrewartha Systems Administrator Data Analysis Australia Pty Ltd

...risk" on Sangoma's page at http://www.sangoma.com/products/p_a101-102-specs.htm; also for E1 and T1, the specs list only compatibility with CAS and PRI signaling, but a pal asked Sangoma directly and they said that the card is compatible with E1/R2 signaling. Cost is also $595 at voipstore.atacomm.com. On the side of the E100P/TE110P there's my wish to support Digium in any way I can for the nice work they are doing with Asterisk; But then, I've been using Sangoma cards for almost 9 years now on data (frame-relay, PPP, etc) applications, and the first card I bought from them is stil...

Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf Seems like a good device, but I can't seem to find anyone actually using them... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061022/6ca85b8c/attachment.htm

hi any chance of making asterisk support these? http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032 -38356249088.htm roy

I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 1 - Grandstream HT-286 1 - Snom 105 Sip phone. I have called and emailed chagres but they have not reply. Nor

Has anyone used either the 8 port or 4 port FXO device from Grandstream? (GXW-4108 or 4104). They seem to be the lowest cost multi port FXO devices that I can find, so I'm getting ready to buy the 8 port version. I just want to see if there are any opinions on the device before I commit to the purchase. If people have not used the Grandstream, are there any issues with using

I -think- that the Sipura 841 was PoE... and I'd been anxious to find out. However, according to Atacomm.com, it's been delayed until mid-January. *sigh* So: does anyone know of a (decent) phone that meets the following criteria, and isn't too expensive? - SIP - two (or more) lines - some form of TCP/IP-based configuration - 802.3af (power-over-ethernet) - 100 Mbit passthrough (not required...