similar to: dynamically modifying the dialplan?

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 =>

Hello, Some of my users have phones lacking a DND button. I need to provide an extension they can dial that will put them in DND, i.e. tell the server not to send them any calls until they get off the DND. I've researched it for almost 3 days now and tried a range of configurations. I'm hoping somebody here has an answer. Currently, I have this in extensions.conf [app-dnd-on]

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

Hello everybody! I've got a big problem: In my small business I use a Home-banking software which I got from my bank. Using it for many years now, it became very very important for me. Needless to say, it's a Windows software. Up to now, it was a Win 3.1 software which ran inside the Windows-Box of my ancient OS/2-PC. (The only reason why I still have this PC runnig OS/2). This summer I

Hi everybody! I've setup my dialplan so that if an extension dials *21*, that extension is added/removed as a queue member to a queue. (State toggled). But it would be great to get an "optical feedback" of that phone's state regarding the queue membership. Does someone know if it is possible to light up a LED under this szenario? Many thanks! Norbert

Hello everybody, I've been using asterisk 1.2 for quite a long time now, but I thought it's time to try a newer version of asterisk. So I downloaded 1.8.5, extracted the tar, ran configure, make, make install ... Everything looks fine (no obvious compile/link errors). But as soon as I start asterisk, it dies with a segfault. I executed asterisk within strace and last action before the

Hello again! after it being "relatively quiet" her for the last weeks, my Astrerisk server was the target of 3 of that nasty REGISTER attacks during the last days. While I can see not much danger coming from these attacks (I use very long, complicated random generated passwords), they are still very annoying, because they always lead to my server crashing. (I think it's some out of

Hi everybody, I've got a very strange problem: As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls. Today I noticed that outbound calls to provider "inode" fail (and inbound from this provider too). On the CLI I get every 20 seconds following messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422

Hello all you Gurus out there! Please could you explain something to me: Currently I try to get ENUMLOOKUP() working. Naturally I do all the testing with my own number. I registered my number at e164.org I paid for registration of my number at a registration agent for e164.arpa (I know, I don't need both. I just did the .arpa registration first and later discoverd the free .org service....)

Hi folks! My bank sent me a CD with the new version of it's Homebanking software. This new version uses internet for communication, the old one dialed with the modem direct to the banks computer. Because the old access will be revoked end of this year, it is very very important to switch to the new version. So I put the cd into the drive, mounted /cdrom an started "wine

Hi everybody, i've been googling for quite some time now but can't find an answer to my problem... I'm using Asterisk 1.2.12.1 with mysql as the cdr backend. In the dialplan i've written exten => 1234,n,Set(CDR(userfield)=blah) exten => 1234,n,Answer() exten => 1234,n,Queue(.....) exten => 1234,n,Hangup() When I'm doing a call I can see that the statement is

Hi Is there any way to set the presence state of a peer to in-use in asterisk 1.8? The idea is to integrate DND buttons on phones to BLF. Regards -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street

Hello everybody, I have a problem and already browsed the mailing list archives but didn't find any help. So I ask here.... My new * Box ist up & runnig. Got access to the SIP server of my Internet provider (Userid, password, phone number, ...). And yesterday I tried my first calls to the outside world. (Internal calls work). Now when I call from the SNOM-360 connected to Asterisk to my

Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Assume you have the messages button on your Cisco phone set to dial 3009. Here's an sample dialplan entry that will make the "DND" and "ToVM" and "Messages" button work as expected. This should work for both -stable and -head. exten => 3009,1,GoToIf($[X${RDNIS} != X]3009,4) exten => 3009,2,VoicemailMain() exten => 3009,3,Hangup exten =>

the caller is out his money anyway when you call any phone and voicemail kicks in, although i think on a payphone they give you a 2 or 3 second window to hang up. Suggest you implement i'm here / i'm away dialplan logic or set the do not disturb button that way when someone calls and the guy is away it hits voicemail right away and the caller can hear this and still have the 2 or 3

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

Hi, On my SuSE 10.1 linux server, I have successfully joined an AD server (which I setup) and am able to create users and groups etc.. on the ADS using the net command. My problem is I cannot (or at least I don't know how) to add/remove users to groups on the AD from my linux box. Is it possible to do this as I can't seem to find any literature on this? I am running samba (and