similar to: Does anyone know how to use the DND feature oc Cisco 7940/7960

Hi all, Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones with SIP. I'm using also op_panel 0.25 (snapshot). I'm using * queues. I want to properly implement DND via *78 and *79. I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd variables and this is fine for FOP. The DND works in normal cases, since I catch it with my Macro dialsip, HOWEVER

Running the latest * CVS and Cisco 7960G and 7940G phones with SIP 6.3 image. I have figured out how to turn on the DND feature through the Settings>Call Preferences>Do Not Disturb - Yes then Save. This puts the phone into DND On and shows a DND image above the far right soft key which you use to turn off DND. There should be a better way. An on/off toggle of the soft key that it

I have put together this call forwarding and dnd config: I'm sure it can be dome with macro's but I couldn't figure that out... anyone care to input. 74 Turns DND on my phone will not ring, drops caller to voicemail... 73 Turns DND off 72+ext forward your extension to another extension and voicemail is left at the forwarded extension. 71 turns off call forwarding. ; dnd Could

I did a clean install to freepbx 2.6.1 and now when i do *76 i get a 1 second flash on the screen then the phone hangs up. the FOP says it is on DND but some ext are still getting calls. once i do a *76 FOP still says I am on dnd. I am running asterisk 1.6.0.21. before i was getting a message like dnd activated and dnd deactivated. i posted this on the freepbx site and here is what i got

I would like to determine through an AGI script if a specific SIP extension has DND on or off. I know that if the SIP client dialed *78 or *79 it is usually enough to just do a: database show dnd to fetch the DND status from the database. However, not all clients dial *78 or *79 (or whichever feature code is defined for DND). Some softphones such as SJPhone have a DND button. When pressed and

Fellow Asterisk Users, I'm running Asterisk 1.2.5, and I've configured basic call queueing using agents. The problem I'm having is that agents who are on DND are still considered (by Asterisk) to be eligible to take calls. This means that calls will hit voicemail even when other agents who are not on DND are available. Any ideas how I can make agents ineligible to receive calls

Hello - I've read Asterisk should be able to activate a do not disturb feature to turn off the ringers on extensions. I checked the wiki and can't find documentation for how to do it. Here's my attempt, added to extensions.conf: [dnd-on] exten => _#78,1,Answer exten => _#78,n,Wait(1) exten => _#78,n,Macro(user-callerid,) exten =>

Only slightly asterisk related I suppose, but hoping someone has attempted this... I have an old installation with a bunch of IP501s, and one died. I replaced it with an IP450, and the user sorely misses his DND button. I hated those DND buttons anyway, as I couldn't control them centrally. I'd *like* to program one of his softkeys to send a *XX sequence to do DND on the

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.

i have posted this question couple of times and never really got any hits i wasn't able to provide any debug info Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309) Verbosity is at least 4 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 == Extension Changed 117[ext-local] new

Is this a problem? What is dnd anyway? Thanks, Jim. May 2 10:44:08 DEBUG[6277] db.c: Unable to find key 'SIP/201' in family 'dnd' -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060502/7bf1f8ae/attachment.htm

I upgraded my 7960 to sip v 6.3 and my display time has now disappeared from the top left corner. Loadid: SW: P0S3-06-3-00 ARM: PAS3ARM1 Boot: PC03M030 DSP: PS03AT38 Here is the section dealing with time in my SIPDefault.cnf file. Does anybody see anything wrong with it or have any other ideas? # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)

Hi, I've noticed that when I set a phone on DND (phone-side DND, meaning it rejects calls with a busy status, SIP 486 response code I believe) the queue keeps on trying the phone over and over again. This creates issues in terms of CDR entries - in a scenario where there is only one phone on DND, and a delay between attempts of 1 second, the queue will attempt to ring the single phone

Well, yes and no. I tested that before and it causes a silent ring instead of a call rejection. I actually want to disable the entire feature. So the phone always rings unless you're actually on the phone. Thanks for the reply though! Regards, Jan ________________________________ Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Jerry

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?

Has anyone come up with a clever way of indicating DND is activated? I've thought of stutter dial tone and using the mwi, but have no idea how to implement these. I'm using Budgetones. My concern is that users will activate the DND, then forget about it not realizing that they are not receiving calls. Thanks, G

Hi, I'm looking to disable rejecting calls from my call center employees. They are using Polycom phones. Is there a way to either disable the reject/DND features on the Polycom phones (don`t think so) or have the Asterisk PBX ignore "Got SIP response 486 "Busy Here" back from 12.23.34.45" response from specific phones/SIP registrations and just keep on ringing?

Typo in your OS79XX.TXT P00 ? instead of P0S !? -----Original Message----- From: Michael L?jtnant [mailto:ml@zyxel.dk] Sent: 17 August 2004 13:31 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released Hi Shaun, Saw you post, and rushed to their ftp-server and downloaded it :-) But, I can't make my phone (7940) upgrade, so maybe you

Hello to all my Asterisk brethren, I have been implementing an Asterisk system for home/office use, and with help from the Wiki and Groups I am doing well. However, with my server and two X-Lite phones I can't figure out how to set a DND or call forward from my extension to another. Probably a real easy thing to do and I am missing something (or a complete section from my

Would anyone know why Voicemail in * doesn't get the DTML keypresses from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do with "dtmf_avt_payload: 101" setting in SIPDefault.cnf in the tftp server? Thanks for any help! Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823