similar to: Polycom IP301: Pass-through ethernet port unusable?

Hi; I'm looking for a Polycom distributor in the UK who can supply a small number (around 20) IP301 / IP501 handsets. Can anyone recommend someone ? jd -- John Daragon john@argv.co.ok argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127

Hi, I am about to buy ip pbx asterisk system but what ip phones do you recommend? Are polycom ip all functional with the ip pbx system??? Be waiting.thanks a lot Marlo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050811/16cc52cd/attachment.htm

I have been testing with Polycom IP600 phones for a month or so. I found out that there are frequent problems with the handset. The problem is I can hear the other end but the other end cannot hear me. I have already downloaded the latest bootROM 3.1.3 and sip 2.1.2 However, there are no problems with the headset or speaker phone. Has anyone encountered such problems before? Thanks.

Hello, I am looking at the Polycom phones. The ip600 has a very nice screen, is that the only real advantage over the ip500 and ip300.. Is it worth the extra dollars? Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: Michael@ITMedic.com.au <blocked::mailto:Michael@ITMedic.com.au> http://www.ITMedic.com.au

Chris Mason (Lists) wrote: > Cory Andrews wrote: >> >> >> >> >> IP430, will sit between the IP301 and IP501, IP430 will have dual >> Ethernet, PoE, and full duplex speakerphone. List price (MSRP) $239 >> street price should fall likely between IP301 and IP501. >> > That looks great, the 301 is almost useless due to the lack of speaker

Greetings, Has anyone noticed that attempting to place a call from the "Placed Calls" list on a Polycom IP501 by pressing the 'Dial' softkey sometimes simply returns the phone to the idle screen? It is not related to the number being dialed, as we have observed two entries for the same number, one of which worked and the other didn't. We've experimented with calls

This feature also works on the IP301 phones. The obvious caveat is that it is one-way only. Still nice for an "all-page" though. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith > Sent: Thursday, February 09, 2006 12:27 PM > To:

I have a Polycom IP501 phone and have set it up to download the config from an FTP server, it did this once and now is in an endless loop of trying to contact the FTP server, failing, then rebooting. When I watch the FTP server logs it looks like the phone starts a session, ends it, starts it, ends it until the phone reboots. It is annoying like nothing I can describe! I have tried Windows 2003

My polycom phone is now hanging at Running "sip.ld". I modified it's config via the web interface to register with my asterisk box. I have tried to restore the default settings wth 468* and it doesn't seem to work. Any ideas? -jonathan

The originating PRI system passes the entire dialed number in the d-channel setup frame, thus the concept of a wait time for additional digits is meaningless. Progressive digit gathering implies that the signalling is occuring 'in-band' as would be the case with DTMF signalling on analog lines. You need to look in the Ascom and find the configuration table that lays out the dialplan for

When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear F@510P) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power

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I am looking at phones for my asterisk system and seem to have a problem. The only Power over Ethernet phones I can find that support the IEEE standard are 3com. Cisco uses its own proprietary ( and is expensive to boot ), snom has a different but equally non-IEEE method, and i'm havent found another phone that I'm confident can do the job for our office. Whats a good high quality ip

Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are

Hi Andrew - > I have a need to be able to identify incoming calls based on some factor > (could be time of day, caller ID, dialed number, it doesn't matter.) -- > Assuming Asterisk can differentiate between the calls I want, how do I inform > the IP501? There are "only" three line appearances -- I can't simply just > ring a different appearance since there

This is my first foray into SIP telephony, so be gentle. :-) The Polycom SoundPoint IP 501 phones have been fantastic so far. I still have a lot to learn when it comes to them, but the manual seems pretty extensive and so far Asterisk has been playing well with them. I have a need to be able to identify incoming calls based on some factor (could be time of day, caller ID, dialed number, it

I am doing some experimenting with Asterisk 1.0.9 and Polycom IP501's. I have the extensions setup, and everything is working well up to this point. Now, I want to setup my system so that a user at an extension can start a recording on demand. I have tried various Google searches, but am coming up empty. Could someone point me in the right direction, or have some sample Polycom/Asterisk

Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? We are still having to touch each unit to enter the ftp server address and password, as well as set many of the options that will not take from the config file. Have a sample config file you are willing to share? What is required in

Hi, Just started using an asterisk-based PBX with Polycom IP501 phones. Am Fairly satisfied and am starting to get into FTP setup of the phones. Have figured out most things except for how button remapping works. In sip.cfg, I have this entry: <keys key.IP_500.31.function.prim="DoNotDisturb"></keys> This works as expected but if I try to change the remapping to any

Hi All - I noticed that the Polycom IP501's are now shipping. Has anyone gotten one yet, and if so, what's different about the phone? Any UI improvements, or is it just better hardware? Thanks, Noah