similar to: Polycom 301's drop last two digits of dialed number

Hi, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example as I have tried various entries for hours now and don't seem to be getting anywhere. Any help appreciated. Kind regards Jamie Heckford Technical Consultant

Hi, We have quite happily been running dovecot with no issues at all for approx. two years now and have had no problem whatsoever. Great work :) BUT (there is always a but!).... yesterday one of our users started complaining that they were having issues with their mail. I checked and although they can still view their IMAP mailbox ok it constantly throws up error messages such as: "The

Hi, Any suggestions to below problem? Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jamie Heckford Sent: 17 October 2006 21:48 To: asterisk-users@lists.digium.com Subject: [asterisk-users] FW: Why does One-Touch record mute/disconnect callif not dialed quick enough? Hi List, Have an odd problem

Hi, Damn I hate upgrades. After updating at the weekend to FreeBSD 5.4-R in a fresh install our samba server just doesn't want to play ball. I have managed to get Samba3 working ok apart from when it comes to using Excel documents. If a user has an excel document open and another user tries to open it, Excel 2003 will fail with 'Unable to open the file, it may be read-only or

Hi List, Have an odd problem with the one-touch record on asterisk 1.2.11. All works ok, however one of our users today discovered if he is a bit slow hitting the 1 key after he presses *, the call seems to stay connected but its almost like it is muted. Haven't figured out the delay yet but it seems to be if the 1 is not pressed within 1-2 secs this occurs. Any suggestions? I tried

Hi, Is it possible to add users/change passwords for users in samba without it being dependant on anything in the UNIX password file? For example I want to add a user to my samba password file without them existing in the UNIX password file. If anyone has any RTFM info they can kick me in it would be most appreciated :) Thanks -- Jamie Heckford Network Manager Trident Microsystems Ltd Tel:

Does anyone have the polycom soundpoint ip's successfully remotely provisioning? I've got the phone pulling default configs, and it's downloading phone specific information, but it's not actually using that information. Any help would be appreciated :) -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Hi all. I just installed bunch of IP450's and everything went well and my customer is happy.... except that they are unable to transfer calls to other extenstions. They can dial them directly just fine. However, when the user is in a call and presses the transfer soft key, they get dial tone, and start typing the extension, say 1008. But by the time they get 100 typed in, the phone tries

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No

Hey gang, I'm hoping someone can help me out here. I've just noticed that on two of my five Asterisk boxes (CentOS 4.4, Asterisk 1.2.12.1), I'm getting the following translation cost for g729: asterisk*CLI> show translation Server 1: g729 - 26 25 25 25 25 24 26 - 53 36 Server 2: g729 - 66 65 65 65 65 64 69 -

Hi, everybody: I have looked at the Polycom entries on www.voip-info.org, and they're outdated and convoluted and full of errors. All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. (The server works with an Xten X-lite softphone.) Has anyone done this? What do I need to do? Thanks,

Is there any way to customize VoiceMail ? I would like to customize the message played to callers sent to the voicemail becouse the extension is busy or otherwise unavailable. Is it a way to record a welcome message and use it ? thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it

Hi, I am attempting to get dovecot up and running on FreeBSD 4.9..... Tried with the default config file and my own one (below) but when trying to login I get the following: Feb 10 17:59:53 cobain dovecot-auth: passwd(jamie): invalid password field '*' It seems like it is trying to read straight from /etc/passwd? I thought it called getpw(). Is this a common error and could anyone shed

I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650. I attempted to simply reuse the existing config files for the old phone on the new phone, but the new phone would lock up on the 4th digit when attempted to dial out certain numbers. So, I downloaded the newest firmware and config templates from Polycom, and attempted to migrate the settings. Seems I'm missing something from

Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying

I am having a strange problem with polycom 501 and dailing. I've read the archives and no one there specifically mentions this problem, so I thought I'd ask here. The problem is that when the user picks up the receiver or pressed new call, sometimes they will enter a number (for example 95072091234) and in the middle of the number the cursor might jump back one digit. So the call above,

I have four Polycom 330 phones connected to an asterisk system. There are other VoIP phones connected too. All of the extensions are four digits beginning with 11. From any of the phones, except the Polycom, picking up the handset to call extension 1103 for example works fine. With the Polycom 330, as I press the second 1 of 1103 it stops taking input and gives me an error. I tried

Hi All, I'm not sure if this is related directly to asterisk or not but on my Polycom 320 when I try to dial a number smaller than 4 digits I get an error on the phone saying "Enter more digits". The dial plan section is listed below. <dialplan dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1"> <digitmap

Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani

Hello, I have a mixture of Polycom SP IP 500 and 300 phones. I have been reading through the administration manual to try and solve this problem, but I do not seem to be able to find the answers to my question. I figured I would ask here and see if anyone has some suggestions. The problem is kind of annoying. After dialing 4 digits, the phone seems to pause and miss the 5th digit, often