similar to: how to determine if a SIP extension has DND on or off

An Asterisk extension calls an Alcatel extension via a PRI link which rings 4 times for about 10-15 seconds and then drops. So if the Alcatel user doesn't answer within 10-15 seconds the call is aborted. (A timeout is *not* specified in the Asterisk Dial command.) It seems however that either Asterisk or Alcatel drop the call prematurely (it's more likely to be on the Asterisk side). What

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

Hi, I'm having an issue with all my calls going out my SIP provider. I'm using a softphone registering to a local Asterisk PBX (I'm using Jitsi by the way - it's great and actively growing). I register as extension 4053 to asterisk server at 10.215.147.115 (alias IP - real IP addr. is 10.215.147.111) and dial a phone number that is routed via an Internet SIP provider. The call

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

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

Hi, I'd like to implement the following: Extension 101 calls 102 but 102 is busy and has no voicemail so 101 is sent to a custom IVR that says something like "extension $EXTEN is $DIALSTATUS. Please try again later or dial $CODE now to notify you as soon as $EXTN is available.". So the "notification" part is what I'm trying to figure out. The extensions are SIP (but

Hi all, I have looked at cisco docs and it says DND is set by pressing the services button and choosing DND. Does anyone know how to configure DND in the services.xml file. I've googled around and not found anything. When you enable it in SIPDefault.cnf it just allows you to use it once. Many thanks for all your amazing work. Daniel Niasoff

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.

Dears my Internet Provider , prevents , sip connections, between sip client(sip phone) and sip server, (asterisk + ser) . both of client and server are mine. is there any solution for tunneling the sip packets? best Mani ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos.

Hi all, i need help to implement a voicemail scenario. What i am trying to do is the following. user X dials a direct access for user Y voicemail and is asked to enter a number (e.g 12345678) and then leaves a message. Then asterisk sends a notification with attachement. The problem is that i need the number entered (e.g 12345678) in the subject. Is that possible. thx in advance.

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

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

Hi, I am running nut with megatec driver accessing ttyS0 as user nut on "standard" kernel (gentoo-sources). It works fine. However, I just built a hardened kernel on a new gentoo machine and have no experience with it. NUT (upsdrv) is failing because it says it doesn't have permission to access ttyS0 even though nut is within the appropriate group. I can add user = root in ups.conf

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?