similar to: polycom 500 help!!

Dean, I appreciate the suggestion. Is it really necessary. I've got slackware already installed on the box. (I consider myself a bit of a Linux guru.), all the drivers, and Asterisk. Everything seems to work fine, EXCEPT getting it to talk to the extension. wouldn't it be easier/faster to just adjust a few settings than to re-initialize my entire box. dean collins wrote: >If

I have working with a polycom IP500 phone. I like the idea of having each line button on the phone as a separate sip device. If I understand it right, each phone could have three extensions (one for each line.) This would be great since I could then use the dialplan to forward calls to the desired extension. I envision something like this: Extenson 101 - Company-A Extension 102 - Company B

Adam, Tor sent this one a little while ago that looks really promising for solving the problem. Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tor Setane Sent: Thursday, January 06, 2005 2:09 AM To: Noah Miller Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re:

Hi All - I'm running version SIP version 1.3.4 on various IP300, IP500, and IP600 Polycom phones. I'm having a tough time with MWI. I thought I remembered somebody on the list saying that they had it working, but I can't find it in the archives now. I have all the phones configured for MWI as specified in the WIKI: ipdmid.cfg: up.oneTouchVoiceMail="1"

I have these very phones and took me a while to figure this out myself. The phone considers each line registration to be a line with a second line. So, call line while someone is on a call and another instance will appear below. That means you only need one registered instance for the phones to get two incoming calls. If however you want to have a second registered extension rung if the first

Hi, Has anyone used polycom phone IP500/600's conference feature? I ran into some issues. Polycom is not behind NAT. Polycom calls pstn number A, (gets two way voice), presses conference, (A is put on hold), calls pstn number B, (gets two way voice), presses conference again, suppose to have all three parties in the call, polycom and B have two way voice, but polycom doesn't have two

What gives with the stupid stupid power system on the Polycom IP500 (I hope they changed it on the 501's). Anyway, does anyone know if you can get some type of joiner to connect two Cat5 cables together, and will this affect the call/signal quality because my Ethernet cable is longer than the dumb 11ft polycom supply (oh and their manual is the wrong about which end goes into the handset

Hello, I have configured my polycom ip600 and ip500. The phone works well. But the clock is wrong and flashes the whole time. Drives me nuts! I have set the time offset on the DHCP / boot server. 36000 (I'm in Australia!) It doesn't seem to help... Mike

We have two small business that run out of our office. One business has 3 phone lines, and the other has only one. In a perfect world, Asterisk would indicate WHICH line (or group) the outsider caller called, so that we would know which way to answer the phone. The incoming calls would go through a voice mail menu first - different for each company. (Press one for sales, two for

Thanks, I got everything to load via ftp. The phone appears to correctly boot from the config files. I also put the latest firmware there and the phone sucessfully loaded it. For some reason, the phone and * don't see each other. This is the part that confuses me. Any clues as to why the phone won't register? -N Jason Brown wrote: > K. Now ere are the configs, minus the

Did you try to boot without lan just the power ... I've had this same problem to and rebooted the device without lan connection -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Louis-David Mitterrand Sent: woensdag 26 januari 2005 11:36 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Hi I am just getting started with asterisk and trying out using the sample files with a Polycom IP500 (latest sip.ld etc) My question: phone status says not registered: /system status/server status "Line 1 poly is not registered" console message says: handle_request: Registration from '<sip:poly@10.1.20.3>' failed for '10.1.20.50' I have tried dialing and

>Message: 9 >Date: Thu, 02 Dec 2004 11:33:16 -0700 >From: "Kevin P. Fleming" <kpfleming@starnetworks.us> >Subject: Re: [Asterisk-Users] Polycom 500, asterisk user opinions? >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <41AF5FEC.7050406@starnetworks.us> >Content-Type: text/plain;

I have sent a Polycom IP500 phone to an overseas remote user who has a Speedtouch adsl modem/router. The phone is connected to the 4 port router which performs NAT for the network. The phone rboots, find the server, downloads it's config and tried to register. Then it immediately becomes UNREACHABLE. Jul 26 17:44:41 NOTICE[21026]: chan_sip.c:9138 handle_request_register: Registration from

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

Greetings, I'm wondering if it's possible to display line breaks with caller ID display. I have the Polycom ip500 phone, and what I am trying to accomplish is instead of the phone saying 'Incoming call from: name/number' i want it to appear on the phone like this Incoming Call from: Menu Context last in Name Number I tried using \n and \\n between the variables (${VAR} \\n

tuning for ulaw g.711 - Polycom IP500 I've got QoS ironed out at this point (dedicated section of bandwidth with VoIP having priority), yet ulaw is still unusable due to small 'holes' in the audio. Running iperf tests shows packet loss at very low percentages (< 0.02 %), and when we do drop, it's only 1 packet at a time. I thought ulaw was supposed to handle this OK.

What's the best price these days for a Polycom IP500 including delivery to NY 10021? Is there an alternative I should be looking at? Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050705/92ffeac0/attachment.htm

Hi I am struggling with hardware choices to get started with. My options are narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G. of importance is: - functionality / integration with asterisk - headset functionality and use - voice quality - build quality Is there much of a difference between Polycom and Cisco? Scanning the group it looks like there may be slightly more

Hi All, I am a remote office that is connected to my office via openvpn on UDP. Voip has always worked well (after discovering g729). Initially I used a softphone, then an analog set on a sipura 2000, then a polycom IP500 (I still LOVE this phone). At that point, I started noticing that the polycom doesn't ring a lot of the time. Since I was desperate for a phone, I didn't upgrade