similar to: Polycom IP301 and 501 with asterisk...

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Jerry Glomph Black > Sent: Monday, January 30, 2006 11:59 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Polycom IP301: Pass-through ethernet port > unusable? > > Have just done a

Anyone know how to override the 99 message limit in voicemail? (yeah, we have a public VM that gets that many a day). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050812/1bff12e4/attachment.htm

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 just implemented asterisk and is such a grate solution...i am using polycom 301 and 501 phones....on lan a iam using g.711 and i have a 16 port linksys switch... the problem come when somebody inside the network is making a call to other extension (in the same network) and is sending an e mail trough internet the quality goes down...it hears rally bad... i am on a

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

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

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

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

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

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, 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

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

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

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

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

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

We just bougth a tdm2400p with all the modules for FXO, but we are having some troubles with the card, cause it aparently is stripping some digits from the dialed number, we tested the same server with a tdm400 and everything worked as expected. We?ve already added "w" before the dialed number with no results, is there any way to solve, is it a bug thanks

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."

Hi I have a trouble with centOS 4 The process kswapd0 eat almost all my memory space, and also this process get almost 100 percent of cpu, and the box go slow I have read that this problem is fix when the kernel is upgrade but I do not know hot to update my kernel. when I type yum list \kernel* I get the following list: [xxx at xxxxxx]# yum list \kernel* Setting up Repos addons