similar to: Re: New to Asterisk and a question

Greetings all, I have been watching Asterisk for a while now, but haven't had the nerve to jump in and start playing until now... I'm fed up with our phone system (or lack thereof) at my office, so I decided to start seriously looking at Asterisk... Mostly as a plaything to get my hands on it, but with the goal of making our phone system here somewhat bearable... We have four

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

Hello all, I've got myself into a bizzare situation that I can't seem to get myself out of... Was wondering if anyone had some advice that might get me 'over the hill' on this... Some background: PBX consists of an Asterisk box (running TrixBox), 4 Cisco 7960's, 2 Polycom IP500's, and now an additional Cisco 7960. The phones are all on a separate LAN. There is

> Hmmm... I have this aweful feeling that I'm choosing the > exact wrong time to ask a "newbie question" :) Oh well, here > it goes. > > The quick question is : "How do I dial an extension?" > (answer is probably - "you don't" in which case:) "How do I > dial my asterisk box?" - I have no outside line, I just want >

Got my first round of IP500s in today. Anybody have any example sip.cfg files they'd like to share? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/a923e094/attachment.htm

Hello, Does anybody have the latest Boot ROMs for the IP500 and IP 600 Polycom phones. I have one of each and can't find the Boot ROM v 3 anywhere to download. I would also love a good sample phone.cfg and sip.cfg files from an Aussie asterisk user to look at. Also the ip500 is having problems trying to load the bootrom 2.6.2 ? Any ideas? Kind regards Michael Felder IT Medic

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

Could someone please recommend a reasonably priced IP phone that works well with *, has a decent (full duplex, echo canceling) speakerphone, has at least two line appearances, and can transfer / conference reliably? The Wiki lists 35 brands of hardphone, but: 1. Most seem to be toys. 2. For many, there is no info on e.g. speakerphone characteristics. 3. When one seems technically promising, e.g.

Hello, I am new to Asterisk and new to this list. I got Asterisk setup and running using Asterisk@home, and purchased a PolyCom SoundPoint IP500 phone to test out. I cannot get the phone to talk to the Asterisk box. On bootup of the phone, it tells me that it cannot contact boot server. Why is that? It gets an IP fine, and I have also tried manually setting the IP of the phone and the Asterisk

Greetings all, My next step is to purchase a nice VoIP phone for my desk. I have a grandstream, and the sound is great, but I'm looking for more of an office style phone, preferably that can handle multiple lines, has a more flexible display (i.e. name as well as number). SIP would be preferable. Any suggestions? Thanks, Eric

>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;

Hi folks, I've got a very bizarre problem recurring when making calls with Polycom SoundPoint IP500 SIP phones and Asterisk. Sometimes when a call comes in to an IP500, one of the sides of the conversation is cut off (i.e. the caller can't hear the callee, or vice-versa). This isn't easily repeated, and rebooting the phone, or restarting Asterisk, doesn't seem to have an

I've had an IP300 for a while now and it's been working fine. I just got an IP500 and when it connects to the FTP server it downloads the new bootrom and says error loading. The bootrom is fine and works on the 300... In addition, I downloaded a new copy to be sure and it still doesn't work. Can anyone give me some advice?

Hi, I just got and setup a new ip500 yesterday and it worked for about 15 minutes. Then it froze during a reboot.Now, when power cycled, the logo comes on for 3 seconds and then the screen is blank and nothing further happens. 468* factory reset doesn't work. I am about to send the phone back, but wonderd if anyone had a suggestion first.

I've been looking at Asterisk for a while now, and want to set up a small installation at my house. I did originally want to get a Cisco IP phone, but due to price and it not being easy to obtain the SIP firmware I have decided again them. I am very interested in the Polycom IP500. I can find many places in the USA that sell it but nowhere in the UK. Also can anyone recommend some good

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

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

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'

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.

Does anyone have any sample configuration files for the Polycom IP500 phones that have been cleaned from the samples that come with the 1.5.2 firmware. I'm wondering how much is necessary, and how much will just work by default. I am really only setting the line authentication , and sntp/timezone-offset for each phone, and everything else is just left as default. Can I just pull unchanged