similar to: Polycom IP601 (mac)-directory.xml changes don't update phone

One more Polycom IP601 question please (sorry for the long intro here to document) ... In order to closely approximate the behavior of the previous telephone system that many of the users are familiar with, I have set up call parking like this: - features.conf [general] section contains: parkext => ** ; What extension to dial to park parkpos => 10-11 ; What

Somewhat off topic. We upgraded a Polycom phone from SIP v1.6.3 to v1.6.6 The phone will no longer send SIP subscription messages for buddies to Asterisk. I have broken the directory file down to make it as simple as possible. Here is what it contains. <?xml version="1.0" encoding="UTF-8" standalone="yes"?> <!-- $Revision: 1.2 $ $Date: 2004/12/21 18:28:05

Hey everyone, I haven't seen an issue quite like mine, so I am hoping anyone who used the Polycom 601's may have an idea. We are going to be switching our office over to Asterisk. All the phones are going to be 601's, I am going to set up a boot server, but for now I am just going to test everything on one phone. My question is I have the phone registered in Asterisk (phone icon

Hi everyone, I have a user that needs a little extra volume on his Polycom IP 601 phone set for all calls (beyond what the volume control currently offers). Is there a provisioning setting for this anywhere? (I'd like to avoid a separate amplifier between the phone and handset if possible.) Alternatively, is there a way to have Asterisk 1.4.x boost the volume to a particular SIP device

Hello List, Not sure if this will be helpful but I made changes to the original Cisco directory.php.txt script and applied them for use on the Polycom phones. This will create an extension directory and alphabetize it based on the sip registrations you have setup in sip.conf. Note that this only seeds the phones and does not synchronize them. Anyway thought it might save people some time. To

Make sure you set nat=yes for the sip user. Asterisk will then send replies back to the source IP address, rather than what's in the Via: header. > -----Original Message----- > From: Warren (mailing lists) [mailto:warren-lists@icruise.com] > Sent: Friday, October 27, 2006 5:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Taking a

I was reading a Polycom brochure and it appears that there is really no special receptionist console and the console is basically a IP601. Is this correct? The only difference is to purchase an expansion module in order to have more shortcut keys for the girls. So, apart from the hardware, as far as the dialplan is concerned, do I just treat the receptionist console as any other extension? Are

Hi, Returning to my office I find two "missed calls" (from autodialers) that my IP601 displays as originating from 01111111111. However the CDR database recorded the call this way: calldate: 2008-04-04 14:18:16+02 clid: 0172752780 src: 0172752780 dst: 2131 dcontext: default channel: Zap/1-1 dstchannel: SIP/0146472131-007a7e80 lastapp:

We are having an issue that I have been unable to figure out how to resolve. I think its related to the Polycom Phones and not the Asterisk configuration, but I'm not positive. We have several Polycom 500/501/601's on both a LAN and at employee homes. The problem we are having is if our internet connection goes down the Local LAN phones loose their connection to the Asterisk Server.

Hey list, Im in this HUGE crisis. Im trying to get a Polycom 601 with two expansion modules to work. I need the XML config files I guess. Does anyone have these I can have? Im trying to get this phone up and running, and haveing MUCHO problems, can someone help me out!! Im sure if I see the configs I can see how it works, just need those XML files!! The ones from the 501 that I have dont seem to

Are all of the sip phones in the same context? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Robert Jenkins > Sent: Tuesday, January 16, 2007 1:44 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [asterisk-users] Polycom IP601 - some hints working,

Received two new Polycom 500 phones. Dhcp and ftp configured properly to load the various files including v2.5.0 bootrom.ld, etc. One of the phones loaded all firmware and config files properly, registers with *, and is usable. The second phone loads bootrom.ld (from the same ftp server on the same wire as the phone), but towards the end of the bootrom.ld load process (about 430 pkts as seen by

I am working on Polycom IP601 console with expansion module. I want to put on the BLF (busy lamp field) feature on all the contact/speed dial names I put on the console but I could not get it to work. *CLI> core show version Asterisk 1.4.13 built by root @ hostname on a i686 running Linux on 2007-11-20 05:26:15 UTC *CLI> sip show subscriptions Peer User Call ID

Hello everyone, Please forgive the exclamation points but I have been battling this one off and on for about four days now. Sorry for the cross post. It all started with a box of IP 501s. I contacted my reseller and obtained the latest BootRom and SIP firmware. Unzipped, configured, copied over to my FTP server (running AstLinux, of course). The phone booted, so far so good. Updated

I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone 0925003705|dns

I'm having problems updating my bootrom to the latest (3.2.2) version. Basically, my phone starts downloading the bootrom, but I am completely certain that it doesn't actually complete the process. It gives me an "error updating bootROM!" message, but it only takes about 2 seconds to complete the download (which, because of my WAN link and FTP server environnement I know is

i recently bought an SP4000 conference phone but having problem provisioning it using ftp, every time it just hangs at "Updating initial configuration..." screen. when i switch it to tftp, it'll work fine. i though it was bootrom/firmware issue so i upgrade it to bootrom 3.2.2/sip 2.0.1 but it makes no difference. any thoughts? p.s. i'm using debian sarge proftpd 1.2.10 and the

I have been struggling over central provisioning for quite some time. I have eagerly watched each post with like problems but have yet to find a reliable answer. I have a Polycom 501 and I am trying to provision from an FTP server, and just to take any routing out of the issue it is on the same subnet. I am running the 2.0.1.0291 firmware and 3.2.2 bootrom. I set the IP info on the phone and

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?

I have a question about DHCP and boot server supporting more than 1 model of Polycom phones. According to Polycom "standards", Polycom phone boots up to get a DHCP address and at the same time getting a boot server string (with username and password) to logon to boot server to download SIP, bootROM and etc. That is okay if there is only one type of phone (that requires a specific SIP