similar to: Polycom 501 Do Not Disturb issue

Hi there! Is there anyone out there using gnudialer? I tried vicidial but couldn't get it to work (does vicidial support SIP trunks anyways?). Gnudialer seems to be simpler, though their web interface needs a little work (version 2.0seems like a step in the right direction but it isn't out yet). How do you register an agent? The documentation is lacking... Jesus Mogollon

I'm attempting to set up my SoundPoint 501 with my Asterisk server. I've configured DHCP and TFTP and successfully updated both the BootRom and SIP application. I've also created a custom cfg file for this phone's MAC address and the settings seem to be taking just fine. I can see that the phone registers with my Asterisk server but 'sip show peers' reports that the phone

I'm running SIP 1.5.2 firmware with a 2.6.2 bootrom on a mix of Polycom 500s and 501s. I'm having a problem with NTP and I'm not sure if it's a configuration issue or a bug in the firmware. I've got the NTP server and GMT offset set in sip.cfg. However, not all phones are in the same timezone. I was under the impression that I could override the offset on a per-phone basis

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

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

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

Hey all, I'll give my reseller a call for support in the morning, but I usually have better/faster luck on the list. I've got a SoundPoint IP500 that I upgraded to BootROM 2.6.2 and SIP image 1.5.2 on someone elses advice, I forgot to change out the old config for the new when I loaded the image up (I guess the config changed a bunch between 1.5.2 and 1.3.1) I was prompted with an error

Greetings all Grandstream released a new firmware and it seems like the speaker phone problem has been fixed. However we updated to firmware 1.0.1.12<http://1.0.1.12>to fix the echo problem but found other problems were now created. The worst of these new problems is that the whole phone starts degrading, the volume starts getting lower and lower. The ringing starts fading and the calls

I've just got a batch of 301s and 501s in and am trying to get them to work. I'd like to manually configure everything via FTP rather than the web or phone interfaces, but I can't seem to find a good source of definitions for all the options in the sip.cfg or phoneX.cfg files. Anyone know of any? Also, I'm having quite the problem getting the Polycom SP 501 to send *any*

This is a dialpaln issue. I solved the same problem recently. For 4 digit extensions you need to append the dialplan statement in the sip.cfg configuration file as follows <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2 -9]xxxT|1xxxT" dialplan.digitmap.timeOut="3"/> Michael > I am not sure what I did but blind transfers do not

Polycom IP501 Assembly: 2345-11500-040 Rev: B Bootrom: 3.1.0.0269 SIP Ver: 1.6.5.0043 I can dial other extensions internally, and can get to voicemail, but when I try an outside number, I hear dial tone, the digits dialed, yet nothing happens when I press "Send". Nothing appears on the Asterisk CLI screen. This phone worked briefly before upgrading from br:2.6.1 and sip:1.6.2

I've got a Polycom IP 600 that doesn't want to DHCP. It DHCPDISCOVERs, there's a DHCPOFFER, it DHCPREQUESTs and a DCHPACK is returned, but 3 seconds later it repeats the process and never boots. The phone works fine with a static IP, and the DCHP setup works ok for an IP 500. I updated to bootrom 2.6.2 and sip firmware 1.5.2 which didn't help. Any suggestions? -- James

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

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

Greetings all I was wondering if any of you have found this problem. I have a setup with 80 GXP-2000s which work great until I decide to stop now my asterisk server. When it comes back up half of the phones (not always the same ones) fail to register with Asterisk. I set up the phones to register and send keep alives (register every 4 minutes, send keep alives every 20 seconds) to no avail.

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

When I try to boot my 501, it runs through the usual stuff, then stops with Config file error Error is 0x4020 and then reboots. The log on the FTP server shows: 0105164151|app1 |3|00|Bootline: ircaIP 0105164155|cfg |3|00|Image bootrom.ld has not changed. 0105164159|cfg |3|00|0004f202f803.cfg could not be downloaded, getting next file. 0105164206|cfg |3|00|Image sip.ld has not changed.

Hi, all. Every now and then, some of my users get "Error" on their phones. A reboot fixes it, but it's quite annoying/inconvenient. I'm running Asterisk 1.2.4, and have the following firmware, etc.: Bootrom: 2.6.2.0032 BootBlock: 2.5.0(11500_030) SIP application: 1.6.2.0041 Any ideas as to why this might be happening? Thanks! -Ken

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