similar to: same extension numbers ??

I have an extension that looks like this: exten => 5555551111,1,Verbose(Door buzzer calling) same => n,Dial(SIP/user1&SIP/user2&SIP/user3) The idea is that any of the three users can answer the phone to let someone in. The problem is that if, say, user2 unplugs his phone then the call immediately goes to his voice mail and the other two do not have the ability to open the door.

Anyone noticed the snom 360 indications are incorrect for US zone? menu->preferences->tone scheme->usa indications.conf: [general] country=us extensions.conf: exten => 1111,1,Answer exten => 1111,n,Playtones(dial) exten => 1111,n,Wait(30) exten => 2222,1,Busy exten => 3333,1,Answer exten => 3333,n,Playtones(busy) exten => 3333,n,Wait(30) hit speakerphone on the

Hello, I am trying to create a dynamic call parking lot using https://wiki.asterisk.org/wiki/display/AST/Application_Park But this manual is not enough to fix my problem : Asterisk keeps trying to park the call in the default parking lot : [Jul 2 11:32:14] -- Executing [3333 at from-770000:5] Set("SIP/testacc77000-00000002", "PARKINGDYNAMIC=parkinglot_test") in new

I'm experiencing some problem using the transfer() application,expecially when a call in received from a queue. I'm using Asterisk 1.4.22.1 This is my scenario: ; this is the piece of code in extensions.conf that place the call in the queue when 1111 is called exten => 1111,1,Answer exten => 1111,n,Queue(2000|t) ;this is the piece of code that calls the user test when 2222 is

All the aforementioned techniques need change everytime on the dialplan. I need the office secretary to edit a file (call file) and place it in a particular folder in their windows PCs. this folder is the outgoing folder of LINUX shared through samba in LAN. i need to make it as easy as possible, please. On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni <mailinglist at linuxista.com> wrote:

Greetings. We are running * on RH9 using a Digium TDM400P four-port FXO card. We use only two ports on the card (ports 0 and 2 in this case). A consultant set this up for us, and it mostly works OK. However, outbound calls use our secondary number rather than our primary number first. This undesirable because of Caller ID; we'd like the primary number to appear instead. Yes, I can

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ

Dear all, I've noticed that when I have a phone registered in Asterisk, and then I register another phone with the same user, the "sip show peers" in the CLI shows that Asterisk replaced the IP of the first phone by the IP of the last one registered for that user. Consequently, if someone calls that user, only the last phone rings!! How may I configure Asterisk to be able to

Hi, I'm having (yet another) problem with R. I have a few data sets that have this sort of format dataset1 ID DATA 1234 value 2345 value 3456 value dataset2 ID DATA 1111 value 2345 value 3333 value What i really want to do is write an R script that says "if the ID of dataset1 and 2 match (2nd row), print out that whole row into a new dataset3". No idea how to do that though.

Hi guys Ive been trying to get this to work for ages now, basicaly im trying to do if ${woteva} = "" (nothing), or its none existenant then do label 1, else label 2. for my last called function, so it will play a different message if theres no last call in the system or it was anonymous. ive tried exten => 1000,1,GotoIf($[${last-call${CALLERIDNUM}} = ""]?4:5) and heaps of

Hi. in my system i have a conference room where someone can call it eg 698 dial the PIN eg 1234 and enter the room as a user. The admin enters in through a different number and PIN. I would like to have a call file and call all participants eg 610-619 at certain time of the day and give them access to the conference. During my try i managed to create a call file where it calls the a SIP phone and

First, let me apologize if this is a simple question, I am trying to get away from Microsoft and learn Linux. In the past, on my Linux test server I have installed samba 2.2.3a and a seperate install of winbind for that same version. I followed the directions I've seen posted and it all works well. I tried to upgrade to 2.2.5 but winbind wouldnt work. So I went back down to 2.2.3a This was

So, I have been testing ekiga 3.0 with Asterisk, and sadly, it don't work. I am told by the ekiga devs in http://bugzilla.gnome.org/show_bug.cgi?id=553595 and http://bugzilla.gnome.org/show_bug.cgi?id=553810 that the problem is that Asterisk does not support SIP forking. The issue is that I have multiple addresses on my workstation: 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500

On Fri, 2007-08-17 at 18:22 +0200, Trixter aka Bret McDanel wrote: > On 8/17/07, Aleks Clark <aleks.clark at gmail.com> wrote: > > Actually, the crazy p2p connections actually reinforce their algorithm > > story. If their p2p algorithms have flaked out, it could cause all sorts of > > trouble. OTOH, I don't think they'd run logins over p2p.... > > >

Per the wiki: Logging off 1. call the extension for AgentCallbackLogin 2. enter your password followed by # 3. when asked for the extension number just press # But if your exten=> is this: exten => 2010,1,AgentCallbackLogin(3333|3044@mycontext) How do they logoff per the wiki's directions? If you use ACBL as above, it never asks you for the extension number because you have

Who would / Who wouldnt need to run SELinux? I have linux server at home. Would I need to run SELinux? What are the advantages of SELinux? What is the average home user doing?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! This is my first message to the list/newsgroup. This weekend and after to have fought by some time with my soundcard with respecto to the voice capture, after assuring to have solved that problem, I installed Asterisk on Debian GNU/Linux Lenny. I made my installation on a KVM virtual machine. In order to begin and according to I could see

Hi, Im having some issues with ghost. Basically my setup is this. Im using a netac U230 usb flash drive which supports FDD booting (emulates a teac usb floppy in a seperate 1.44mb partition), ive installed syslinux onto this (as the t61's i was testing this on wouldnt boot a normal dos image). From this, i have memdisk loading a windows ME based dos boot disk that to a netware network

here's an idea for icecast.. the ability to create a relay dynamically by passing a url to the admin interface. (instead of having to add to the config file and kill -s HUP) such relays wouldnt be permanently stored in the config file (unless icecast starts writing to its config file) but would function as long as the server stays up, or until stopped thru the admin interface. --mark

Is there a limitation to the number of variables you can set from a PHP agi script? I have a simple example and I can't get it to let me set more than 1. I am pretty sure I am just missing something, but I've searched all over an can't find the answer. Here is the extensions.conf part: exten => _XXXXXXXXXX,1,AGI,diallocal.agi exten => _XXXXXXXXXX,n,NoOp(${ISLOCALCONTEXT})