similar to: Reboot any(?) SIP Polycom -- provisioned or no.

Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble on registration(? -- maybe it's on acquiring an IP?) has started again. I still have the old sip.cfg, but can't figure out which option it is. Any help? Thanks! -Ken D'Ambrosio -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html Howdy, all. What's the difference between "split" and "combined" firmware, which can be seen at the above link? I've googled to no avail, I'm afraid. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi, all. I want to have a button on my receptionist's 601 that, when pressed, will forward her current call to a given extension. Is there any way to do that? I've tried to RTFM, but I'm coming up empty. Thanks, -Ken D'Ambrosio

Hi, all. I see that the Polycom SoundPoint IP 330 supports VLAN... but I don't quite see how that works. Do you point a non-VLAN'd segment at it (akin to when you uplink a VLAN_enabled switch), and have the phone implement the VLAN? Or...? *puzzled* Thanks much, -Ken

Hey, all. I'm all over MWI, but I gotta say that I think the Polycoms go a bit over the top. The blinking LED is enough for me; how do I disable the stuttered dialtone and the audible warble? I've looked through the config files, but there are a HELL of a lot of options, and I haven't been able to find those particular ones yet. Thanks! -Ken -- This message has been scanned for

I can't figure out how one upgrades the boot ROMs for the Polycoms. I've read the wiki, I've checked the docs, I've looked here, there, everywhere; I hoped it might be through the web interface, through DHCP... but I can't see how to do it. Suggestions? Thanks! -Ken

Hey, all. I've got an office set up with Asterisk, and forwarding's got a bit of a glitch: When they forward, they listen for the remote phone to ring, then hang up. If the remote phone doesn't connect, it goes to the original phone's VM. Is this Polycom's "fault," or Asterisk's? I've been reading up on blind/supervised forwards, and, honestly, have

Hey, all. I quoted a customer about $100 for some cheap SIP phones. I was planning on using the BT-102's, but he called said they look like "Princess phones," and I have to admit that he has a point. Some of the other inexpensive phones look decent, but (for example) the SPA-841's wiki entry says the remote end gets a lot of static. Since it'll be being used from a noisy

Hello all, I need the bootrom.ld file to set up some Polycoms I have Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A I've publiched on my FTP files downloaded from http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html (3.2.3 combined and split zips) but my phones are still showing the message: "error, application is

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:

I`m trying to provision my Polycom 501s using Pure-FTP. Everything has been working fine, until I put a new bootlog.ld file. Sniffing the packets between my phone and the FTP server during transfer, I noticed that the file is not 100% sent (it seems to send about a third of it, and then the phone wisely detects it as being corrupted and reboots). I have suceeded in using other FTP clients to

With my thanks to Brian West and his offering in the thread, "Subject: Re: [Asterisk-Users] Remote reload Cisco 7960" I offer PolyReboot.pl, a perl script for rebooting Polycom IP Phones PolyReboot.pl takes an IP address as a single argument and reboots the phone. You must have a cfg file in the Polycom style, i.e., 00ab00cd00ef.cfg - all lower case. Further, you need to use ftp for

Hi all, I saw that there was an auto-provisioning feature on asterisk 1.6.x for the Polycoms. But no real documentation. I would like to know how, exactly, does the network has to be configured to allow that. I used to provision my Polycom phones with the help of tftp or ftp. But if there's a way to do that directly with Asterisk, I would prefer that. So I need to know what I need to do

Kevin wrote: > There is a reboot script posted on the wiki to reboot Polycom > telephones. When I execute this script, I get the following messages. > I am concerned as this is causing issues with asterisk and the PRI. > Does anyone have any ideas why this would be happening? > > > > asterisk console: > > -- Remote UNIX connection > -- Remote UNIX

Guys, this is a weird question but has anybody disabled the blind button that appears on polycoms or know if you can disable the use of blind transfers on polycoms to make any transfer attended? Thx!

Hello list- I'm having an extremely odd issue with an installation of mine. The system is running * 1.2.12.1 and currently handles around 100 handsets. With the exception of a few Grandstream DTA's, all devices are Polycom 320, 430, or 601's. After a recent power outage, I'm having an extremely odd issue with one of the handsets. One of the Polycom 601 units simply reboots every

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

Hi everyone, I have been dealing with a certain issue with a particular customer site for months now. The problem occurs when there is an error with caller id as shown in the following: WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error on channel 'Zap/3-1' When this happens, it appears that the call still goes through as I can see the caller still navigating

Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones....

I have provisioned a bunch of Polycom 301 phones to get the config files from my ftp server. Out of the 4 phones 2 get the config file however the other 2 cannot contact the boot server. I have reboot the phones a number of times remotely (the client is 400 km away) through vnc and logging onto the web config internally. No matter what I change on the web config page it is not saved. I feel I