similar to: Newbie Polycom: Will a big 0000-directory.xml crash the phone?

I am using Polycom IP600 phone. If I call a phone which has DND (do not disturb) enabled, the message to the caller will be "The person on extension ... is on the phone, please leave a message ...". Is there a way to pick the "person ... not available" message instead?

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

When I downloaded the sip and bootrom from Polycom website, I noticed a file called SoundPointIPWelcome.wav. However, I have no idea where and when it was used. I played the wav file but I have never heard the phone using this wav file before. Does anyone know what it is used for?

Hi Polycom experts, I'm having a problem getting changes to the Polycom IP 601's (mac)-directory.xml file to update the button list on the phone. If the phone is newly provisioned (i.e. if I "Format File System" on the phone) then the new list will show up on the buttons, but of course this is pretty drastic way to do it. - Environment: Asterisk test setup with 7 phones,

I did not really spend too much time on looking at call forwarding and wonder if someone could help me. It seems that for setting call forwarding on the Polycom phone itself, only "forward all calls" will work. The other call forward function like "forward if no-answer for n rings" or "forward if busy" does not work at all on the phone. If that is the case, it

Hello all, We have an Asterisk server that has worked without issue for a while. Before, only Sipura and Polycom 500 series phones were used. Recently, we've added a few POE switches and 20 or so Polycom 330's. The 330's seem to lock up often. One easy way to do this is by hitting the * key. I looked thru the XML files for dialplan and what not, nothing stood out.

Hi, I have a strange problem with all of our Polycom 550 & 650 phones. I am running a TFTP server on my Asterisk server and option 66 Boot Host pointing to Asterisk on my DHCP server. The auto-provisioning is working because the phones are registering correctly with their extension. If I change the MAC.cfg file to another extension and reboot the phone, it will reflect the new ext. The

Calling all Polycom gurus: I am using Polycom IP601 phones with Asterisk 1.4.21.2 In all Polycom phones, I set the following in sip.cfg. <dialplan dialplan.impossibleMatchHandling="2"> </dialplan> (I leave the digitmap unchanged because I thought setting impossibleMatchHandling will ignore the bitmap) ...so that I could dial any number by entering a variable-size

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.

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

> >> Jason Brown wrote: > >> | Anyone have experiece with polycom phones? > >> | > >> | I am experiencing a really weird problem. In an office > where I have > >> | the following extensions: > >> | On the Polycom phones, when I want to dial from extension > >> 100 to any > >> | extension 120 or above, or dial out, it

Anyone have experiece with polycom phones? I am experiencing a really weird problem. In an office where I have the following extensions: 100 101 102 103 104 110 111 120 130 140 141 150 200 On the Polycom phones, when I want to dial from extension 100 to any extension 120 or above, or dial out, it dials just fine. If I want to dial from extension 100 to extension 101,or 102 or 103 or 104,

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

> > Jason Brown wrote: > | Anyone have experiece with polycom phones? > | > | I am experiencing a really weird problem. In an office where I have > | the following extensions: > | On the Polycom phones, when I want to dial from extension > 100 to any > | extension 120 or above, or dial out, it dials just fine. If > I want to > | dial from extension 100 to

I setup a number of remote phones on public IPs using the web interface. Now my question is how do I change the default Polycom:456 password via the web interface. Is there a hidden way or does it have to be done via FTP TFTP? Thanks, Steve Totaro

I know that Cisco 79xx phones support XML applications being launched from the screen, and I've seen that the Polycom phones claim to have the same feature. Before I throw myself into the rat's nest of learning how to do it, does anyone have any sample XML applications for the Polycom phones, specifically integration with Asterisk (though any samples will do)? The admin guide from

Hi all, I'm having a problem with a polycom 301 not picking up a ZAP call. Below is the CLI output of the call. I have: TDM400 with 2 FXO lines Asterisk 1.2.14 Polycom 301 When I dial the first ZAP line, I choose an extension that rings the polycom, polycom rings and I can pick it up and the call is bridged. When I call my second zap line, the polycom rings, but I cannot pickup the

Hi, Has anyone used polycom phone IP500/600's conference feature? I ran into some issues. Polycom is not behind NAT. Polycom calls pstn number A, (gets two way voice), presses conference, (A is put on hold), calls pstn number B, (gets two way voice), presses conference again, suppose to have all three parties in the call, polycom and B have two way voice, but polycom doesn't have two

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

Hi We have polycom 430,501 and 301 phones. Our customer does not need DND feature in any form. I can disable this feature from asterisk server but How can i disable this feature on phones. In the sip configuration file i found the parameter that change the phone behaviour during DND from busy to normal but still if the phone is in dnd mode the phone ringer would be off which is unacceptable.