search for: ip501s

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

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

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

The answer is yes, I think, but I don't recall precisely how off the top of my head, and I'm walking out the door in a moment. The phone will hold more than a dozen distinct ring tones which you can create for yourself, and you can have asterisk direct it to use a ring tone independently of line appearance. The most direct way would be with the SIP Alert-Info field, but the phone itself

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

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

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

Greetings, We are trying to make our corporate directory (around 400 entries) available via TFTP to some Polycom IP501 phones. A small (~40 entries or so) file works, but the full file fails to load. Does anyone know what the upper limit on directory entries is? The size of the XML file itself is only 60K - you'd think that would all fit into the phone with no problems..... I would

BOFH told me he uses it to listen to his co-workers > -----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: asterisk-users@lists.digium.com > Subject: SOLVED: Re: [Asterisk-Users] Polycom IP501 with Asterisk - > distinctive

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:

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 gang, I know this is off-topic for Asterisk, but I don't know where else to ask: I've setup a central directory.xml file for my Polycom IP501 phones with a list of all the internal extensions. None of them have <sd>1</sd> as I don't want to enable any speed dials, just have a list in each phone. However, when a phone boots, it seems to pick a random entry and put it

I remember seeing something like this on the list a while ago, but I'm darned if I can find it. We have a number of Polycom IP501 phones, some of which have more than one registration on them. When a voicemail is left for a phone with only one registration, the MWI lights up and stays lit until the voicemail is listened to. However, on our phones with more than one registration, the MWI

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

Has anybody got the polycom acd function to work? I have the following setup: Debian 3.1 - 2.6.8 linux zlib-1.1.4 libpri-1.2.3 zaptel- 1.2.6 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one error when doing a make install about needing a newer version of libpri and zaptel, I got the above versions from asterisk.org, are there newer version anywhere else? In the sip.conf

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

Greetings, The Polycom SIP 1.5 Admin Guide says this: "3.1.8 Connected Party Identification Where possible, the identity of the remote party to which the user has connected is displayed and logged. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion."

Using FTP to configure 501. Gets past "Running... App = sip.ld" and: Welcome! Processing configuration... "This may take a few seconds." Then it displays: Config file error Error is 0x4020 and reboots continuously, repeating the above. Anyone seen this before? Is this a corrupt *.ld file? An FTP error? (PASV)? A missing SETUP setting? Thanks! -- Jim Rice by Design

Hi, Has anyone had any great difficulties with QoS using the second ethernet phone in these Polycom phones for desktop machines in a converged network? I had heard that these can cause difficulties when used in this manner. I have always tried to persuade customers to go with 2 ethernet drops per workstation to avoid having to use the phone as a switch. I apologize for this question not being

Is anyone using fixed Park buttons (some of the ones on the left side of the screen) on a Polycom phone? Here's what I mean: - Call is received and parked, by the user pressing an unlit park button (e.g. 701) and the call is parked there. - The call can be picked up at any other extension by pressing the flashing park 701 button. - Once the call has been picked up, the 701 park slot is idle