similar to: Polycom ip301 hangs at Running "sip.ld"

Hi, I am about to buy ip pbx asterisk system but what ip phones do you recommend? Are polycom ip all functional with the ip pbx system??? Be waiting.thanks a lot Marlo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050811/16cc52cd/attachment.htm

> -----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

I'm getting these messages every 7-10 seconds. -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP '532' at x.x.x.x port 56988 expires 60 -- Registered SIP '529' at x.x.x.x port 51444 expires 60 -- Registered SIP '529' at x.x.x.x port 64044 expires 60 -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP

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

Yeah....I think that every install I have done the first thing that happens is "why is there a delay before the call connects?" and the answer is "you have to hit dial or wait 10 seconds". -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Rymes Sent: Thursday, August 11, 2005

Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the XXXXXXXXXXXX-directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in

Polycom Provisioning Tool Updated. I made a bug fix that was reported, which was causing the directory creator to not work when there was an invalid character in the filename of the csv. I have also posted an FAQ: http://www.wintrisk.com/ppt.html#FAQ Download the new one, and tell me what you think! It's free, and mildly useful! http://www.wintrisk.com/ppt.html Yours, Michael Munger,

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

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.

How do you guys like the 330 and 320? I've been looking at this as my "standard" phone, since it's relatively cheaper than the 501 which is the phone I currently push with my PBX systems. Most of my customers do not use more than one line per phone, so having 3 lines on the 501 is not necessarily useful. Also, I read that the phone offers TLS security. What does that mean?

nat is equal to yes, and server definition in ftp provisioning server is correct. i followed packets between phone and asterisk, it seems for some reason asterisk is not happy about challenge response its getting from polycom. why its not happening in the same LAN, beats me! and also NAT device is cisco SIP aware and works flawlessly with Linksys. the only close definition of issue i have found is

Hi all, Has anyone made up custom ring tones for the Polycom SIP phones? We use different rings for different lines, but the ones it comes with are all very similar. In the interesting of sharing, here's one I made up for paging: <PAGE_BEEP se.pat.ringer.13.name="Page Beep" se.pat.ringer.13.inst.1.type="chord" se.pat.ringer.13.inst.1.value="12"

Hello, We have a problem on a recent asterisk install with Polycom 30x phones; Sometimes (can not reproduce or find the logic of the problem after one week one analysis), the called party (even incoming or outgoing call) can not hear the calling party, as other flow works (caller hears called). This occurs between 5 and 10% of the time. The configuration is the following: - Asterisk 1.2.9.1 -

The only phones I have much experience with are the sipura spa-841's, the netweb 301/302 phones (which I really don't like) and the polycom 300/301's. It applies to the sipuras and the polycom's for sure. I can't remember about the netweb, we quit using them sometime last year. -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Hi all, When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has resulted in a MWI clearing delay of around 5 minutes. After listening to a voicemail and deleting it, the Polycom VVX 601's MWI light is left on for around five minutes, before clearing. Installing Asterisk 13.24.1 did not fix this. Moving back to 13.23.1 allows the MWI to clear immediately. I see a note in

You are right. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tarpo, Louie Sent: Thursday, August 11, 2005 5:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone You write out a

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

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 am getting the following message when trying to lookup up a number via Dundi: Feb 1 13:39:24 NOTICE[20146]: pbx_dundi.c:1309 update_key: No such key 'office.pbx.bluegrass.net.pub' for creating RSA encrypted shared key for '00:a0:c9:55:91:89'! I have created keys on each box with "astgenkey -n office.pbx.bluegrass.net" using the host name for each box of course. I

Below is a snipped debug on our PRI. We are getting number only for the CallerID but the telco says they are sending us Name and Number. We are getting the Name in a second frame but Asterisk is not passing it to the device it rings. The message below says "Presenation allowed of network provided number" which leads me to believe Asterisk thinks it should not be displaying it. Can anyone