similar to: New VM feature: broadcast and delete=yes

The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox

Hello I assume it's possible to do this with Asterisk: To train a new worker who works remotely, I'd like to have him listen in on support calls so that he gets to learn the kind of problems that come in, and how they're solved. When a call comes in and the support person thinks it's worthy to have the trainee be part of it, he will ring the trainee so he can join the call.

Hello everyone... I'm trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm)

I was asked today if we could somehow have a trainee on the phone with a supervisor conferenced in, but somehow have it so anything the supervisor says is only heard by the trainee and not the customer. Is there a feature like that?

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my

Here is what I have now. Where should the line " exten => _.,1,macro(record-on,${EXTEN},${CALLERIDNUM})" go should it be under [sip]? Right now if I call sip to sip monitoring starts and the calls connect but I only get 44 byte files. If I call and iaxtel number monitoring starts but call never gets placed and again 44byte files with nothing in them. Thanks for the help. [iaxtel]

Hello All, In reviewing some code in file 'bsd-cray.c', I found a possible issue where data in the following code may not be properly scrubbed in the case IA_BACKDOOR in function 'cray_setup', which is below: case IA_BACKDOOR: /* XXX: can we memset it to zero here so save some of this */ strlcpy(ue.ue_name, "root",

okay, im far from a pam expert, but i don't see any mention of winbind there? It's my weekend at the moment so i can't get to my test box at work to show you my pam module using winbind, but maybe you should check out this page on my website, it's using ldap try and use this and replace any mention of ldap with winbind http://www.yourhowto.org/content/view/35/9/ or i have a

I have this situation: class Employee < ActiveRecord::Base belongs_to :designation end class Designation < ActiveRecord::Base end I do the following at the irb console: Step 1: Find an employee >> emp = Employee.find 3 => #<Employee:0x35a7d34 @attributes={"designation_id"=>"3", "id"=>"3",

Heya, I'm a trainer for Red Hat, preparing to teach a course on nifty directory services, and have been doing a little research on related matters. I stumbled upon last weeks thread re: group policy on Linux / Samba machines and thought you folks might be interested in the following. A company called Nitrobit has a group policy implementation that works in conjunction with Samba and any

Chaps, I recently added an incoming VOIP account to my asterisk box. When the PSTN number associated with this account is dialled, the call rings once and then asterisk starts playing music on hold, even though all the extensions continue to ring. Variations of answer() and ringing() don't seem to help. I'm sure I'm missing something spectacularly obvious, but the wiki and googling the

I want to record all phone calls made inbound and outbound. I'm new so having a hard time getting this started. Here is what I have so far but isn't working. Can someone help me out? Thanks, [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) [sip] include => macro-record-on include => iaxtel exten

Hi all. I'm trying to get a simple configuration working so I can later expand it to something more interesting. I'm using kphone to call an extension on the * server. When I try to connect, I get this error: DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on

I have a T100P card connected to an Adtran and then a T1. I have added the following configurations to Asterisk...but, when I dial 9 and then a local phone number, it bounces between the dial tone and silence and the *error* light on the Adtran blinks. zaptel.conf span=1,0,0,esf,b8zs fxsks=1-8 loadzone=us defaultzone=us zapata.conf [channels] context=from-sip signalling=fxs_ks

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040715/b37cee4b/attachment.htm -------------- next part -------------- ? Hi friends I have some doubt in connecting my firefly3rd party softphone from windows machine to asterisk server in linux . My asterisk is behind the "Port Restricted NAT". I am using STUN server to cross the

Dear R-help, Please could I have some quick guidance on what I'm doing wrong when trying to instal R software? (I have read the R-FAQs and instructions, and watched youtube instructional videos on installing R, but they didn't help) I've attached screenshots to hopefully make what I've done clearer. Basically, R doesn't seem to be installing correctly and I can't figure

While building OpenSSH 5.2p1, "make tests" was failing on my system with the error "no sshd running on port 4242". After much head scratching, cursing and rooting around in the test scripts I finally figured out the real cause... something is already running on port 4242 (in my case, the Juniper Network Connect client). This got me thinking, it might be nice if the code

I installed a Centos 4.x system using a lvm install across four HDDs. It is my first install using LVM. System had a power-failure and stopped booting up. A new trainee simply took out the HDDs and restarted the file-server on a fresh HDDs. Now the problemis that the four HDDs have data. But the order of the HDDs (of install....1st primary, 2nd primary etc.) is unknown. Earlier we used to boot

Hi, I have a bug and I don't find it. Can u help me please. In attachement : - My script file : CPU_study.txt - My data sample file : a.txt When I use it, the error message is : > source("H:/R/_workspace/CPU_study_1.R") Erreur dans "[<-"(`*tmp*`, i, k, value = numeric(0)) : rien ? remplacer (= error in ... nothing to replace) I try a lot of thing but it

ive got asterisk running with chan_sccp and three cisco phones (2 7910's and 1 7960). lots of bugs. when i press the speed dial button on either 7910, asterisk dies. also, if i dial from the 7910 to 7910, everything works fine. i can dial from or to the 7960 once, and then one of the 10's and the 60 die and try to reregister. if i take the 7960 out of the mix and remove its