similar to: "Sticky" Polycom 501 keys and handset

I have the following situation: Asterisk 1.2.1 25+ Polycom 501 telephones. Bootrom 2.6.2.0032 Application 1.6.2.0041 Some 501's local to my network, some across the great INTERNET divide. PRI connected to Sangoma card. Issue: horrible echo (and squeals, and "underwater-like" sound) on speaker phone when calling from extension to extension. echo not present when calling outbound

I'm attempting to set up my SoundPoint 501 with my Asterisk server. I've configured DHCP and TFTP and successfully updated both the BootRom and SIP application. I've also created a custom cfg file for this phone's MAC address and the settings seem to be taking just fine. I can see that the phone registers with my Asterisk server but 'sip show peers' reports that the phone

Somebody was helpful enough to give me the very latest release of Polycom's firmware (2.1.0). Unfortunately, I still get that issue. So I'm stuck asking again: Anybody ever got that? Mike _____ From: Mike [mailto:list@virtutel.ca] Sent: Wednesday, April 11, 2007 13:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Polycom 501 issue with latest

Seeking recommendations on handsets for use with Asterisk. I've been looking at the Aastra 480i CT because of its cordless handset and also the new Linksys SPA-942. Anyone using either one of these with comments on them? Any other thoughts on good reasonably priced handsets? This is for just a couple of people who work from home offices and will be connecting to an Asterisk server hosted

Hi, I've upgraded a few Polycom 501 to SIP 2.0.3b (can't get 2.1.0 because of Polycom's firmware policy, but this is the "latest publicly available" from Polycom's web site). I've noticed that some keys get "sticky" though. Soft buttons for example (i.e. "end call") need to be pressed 2-3 times for them to react. I've downgraded to

Hi, I have a Polycom 501 that is currently unusable because I started a firmware and sip upgrade that I can't complete. The Ubuntu box address is set static at: 192.168.1.101. The phone address is set static at 192.168.1.51. The phone settings for the server menu are: Server Type: Trivial FTP Server Address: 192.168.1.101 Server User: PlcmSpIp Server Password: PlcmSpIp (not sure what it

I've been noticing that my group of Polycom IP 501 phones seems to randomly reset themselves nearly every night (I guess it usually happens at night, since I've never seen it happen while I've been at work during the day).. When I say "reset", I mean, the hands free volume and ring volume are set to the default and the call logs (received calls, missed calls, placed calls)

I have a couple of Polycom phones, bootrom 2.5.0, SIP 1.3.1.0056. Works great with Asterisk when I power on the phone. However, after some time, say an hour, I cannot receive calls on this phone. On Asterisk, when I do "database show", it does show the phone in there, but it cannot reach the phone. Is there a way to keep alive the connection with Polycom? Any suggestions? Thanks, --

I have the following setup in my test lab (which reflects very much my production installation, just on a smaller scale) Asterisk server ------------- Internet -------------- Home router (Linksys) ---------------Hub ----------------> Polycom 501 (Phone A) |-------------------> Polycom 501 (Phone B) All calls go through my asterisk server, even if its from one Polycom to the other. If

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-----

Does anyone know where I can get replacement handsets for the Polycom SoundPoint IP phones? Or does anyone have any they want to sell? From the looks of it you have to buy a whole new phone to get a new handset. My vendor, TriaTechCOA, told me I had to buy a whole new phone to get a handset, which is pretty ridiculous. Maybe there is a more sane vendor I should be buying from? Thanks, -Ryan

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

Docs? Polycom has docs? Where would one find this fabled land of... err I mean Polycom does stating what ftp servers are supported? Doug. -----Original Message----- From: Ken D'Ambrosio [mailto:ken@jots.org] Sent: Tuesday, March 07, 2006 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

...even when I do the "factory reset" (4-6-8-* then 456). I tried using FTP and TFTP, but even though the phone uploads the log, I get these errors: 0927211350|app1 |3|00|Time has been set from 0.us.pool.ntp.org(69.60.124.59). 0927211350|cfg |4|00|Could not get all 512 bytes of the header. 0927211351|cfg |4|00|Could not get all 512 bytes of the header. 0927211422|app1 |4|00|Loaded

Hello everyone, Please forgive the exclamation points but I have been battling this one off and on for about four days now. Sorry for the cross post. It all started with a box of IP 501s. I contacted my reseller and obtained the latest BootRom and SIP firmware. Unzipped, configured, copied over to my FTP server (running AstLinux, of course). The phone booted, so far so good. Updated

While I'm asking about the Polycom ip500, the answers for all phones where mic/handset/headset levels are adjustable would be of interest to many I'm sure. For the ip500, the default value for the handset seems to be voice.gain.tx.analog.handset="3" I've noticed that echo all but goes away when one reduces the mic volume on almost any phone. My question is, for you users

Hello all, We have an Asterisk server that has worked without issue for a while. Before, only Sipura and Polycom 500 series phones were used. Recently, we've added a few POE switches and 20 or so Polycom 330's. The 330's seem to lock up often. One easy way to do this is by hitting the * key. I looked thru the XML files for dialplan and what not, nothing stood out.

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.

Hi I am looking for a good wirless headset to use with the Polycom Soundpoint 501 phone. I would greatly appreciate hearing from anyone with good experiences with a specific device. Thanks -- Jim Freeze

I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone 0925003705|dns