similar to: Transfer sounds - notifications

Dear List, I ran into some problems with time-series-Data. Imagine a data-structure where observations (x) of test attendants (i) are made a four times (q) a year (y). The data is orderd the following way: I y q x 1 2006 1 1 1 2006 3 1 1 2006 4 1 1 2007 1 1 1 2007 2 1 1 2007 3 1 1 2007 4 1 2 2006 1 1 3 2007 1 1 3 2007 2 1 I am looking for a way to count the attendants that at least have

Hi, Could anyone tell me what is wrong: > length(unique(mydata$myvariable)) [1] 2 > and in t-test: (...) Error in t.test.formula(othervariable ~ myvariable, mydata) : grouping factor must have exactly 2 levels > I re-checked the code and still don't get what is wrong. Moreover, there is some strange behavior: /1 It seems that the error is vulnerable to NA'a, because it

Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

Hi all. I am trying following scenerio for call park & pickup. voice is flowing established between B & C, after call-pickup ( instead of A & B ). can anyone please clarify why it is happening like this, ( or ) do i need some more configuration for park&pickup ? A B

Hi, I'm trying to get a sweex ups working with nut on debian sarge. This one: http://www.sweex.com/producten.php?lang=1&%20sectie=&item=59&artikel=95 It's shipped with UPSmart and according to the compatibility list it should work with the genericups upstype=7 driver. But it doesn't. I get: mybox:/etc/nut# /etc/init.d/nut start Starting Network UPS Tools: (upsdrvctl

Hello, have some of you tried sweex UPS 1500VA (http://www.sweex.com/en/assortiment/power-security/power-protection/PP220). It is not listed in hardware compatibility list. As I can see, sweex can be problematic with drivers, but it is not possible to change this UPS to other model now. It is connected by USB and lsusb says only: Bus 002 Device 004: ID 0665:5161 Cypress Semiconductor USB to

Dear Charles and team Just thought to share my findings getting a Sweex PP200 650VA UPS working with NUT (I didn't see this particular model listed on the HCL.) I picked-up one of these units quite cheaply at a Maplins store (before the chain closed-down) as a second UPS for my home/office - the first being an old Zigor Ebro650 - a long story for another post... NUT v2.7.4_7 is running

Hello folks, I have a question regarding a sweex product. Its bundled with UPSmart v 2.12. I installed nut accordingly to the user manual included in the software. A quick check of data/drivers.list revealed to me i should use the: "Sweex" "500/1000" "contact closure - shipped with UPSmart" "genericups upstype=7" driver this driver

Hello, I switched from my PSTN provider to a voip provider. (Voicedata in the Netherlands) >From the moment i switched all inbound calls are terminated after aproximatly 1 minute. The provider tells me it's not their issue since I have no other configuration than all their other users. What can I do. I removed all asterisk functionality by forwarding the inboud call directly to a local

Hi All Apologises if this has been disussed and I missed it. My SetUp I have a sip phone registered to an asterisk box (a1) in one location 1. This phone dials an extension which is in another location, so a1 passes the call via IAX to the other asterisk (a2) in location 2 which then dials the local phone. My Problem The caller ID setup in the sip.conf for the phone registered to a1 is not

Hi all, I am probably missing something ultimately obvious, but I have a problem configuring freepbx... Using Edgy Eft with the cvs freePBX 2.2.1 and followed the Ubuntu installation guide on freepbx.org. System pxe-boots from a server with NFS root on same Using * 1.2 current (from source, not .deb's) Using mISDN-streams (from source, not .deb's) Using freePBX-2.2.1 (from source, not

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the

Hi all. When a user transfers a call by pressing the "transfer" soft button on their phone, I'd like it to "beep" at them when the transfer is complete. I've got it turned on in features.conf: xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr ; to indicate a failed transfer However, it seems that

Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext => 100 parkpos => 1-5 context => parkedcalls parkingtime => 100 transferdigittimeout => 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer => *2 blindxfer => # disconnect

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring

I work for a call center and we are looking at using asterisk to have our operators take calls. Our message taking software records all the calls on the operators computers. Right now we use these recording controls from radio shack that plug in between the wall jack and the phone and plug in via a 1/8 inch stereo connector to the mic input on the computer. If I buy an IP phone I can't do

Does this require a new entry in scripts/hotplug/* and scripts/udev/*? -- Peter Alexander Gordeev wrote: > > Author: agordeev-guest > Date: Sat May 26 12:04:26 2007 > New Revision: 924 > > Log: > data/driver.list: added SVEN Power Smart RM 2000 to the list. > > Modified: > trunk/ChangeLog > trunk/data/driver.list > > Modified: trunk/ChangeLog >

I am trying to move from IAX2 to SIP for voipbuster, moving at the same time to sip1.voipbuster.com. When I try calling out, I see that there is SIP exchange, and in many cases also RTP data being exchanged. Hover in a very large number of attempts the connection is not established. Half of the time there is no RTP, the rest of the time there *is* RTP data flowing in two ways, but no ringtone is

Hang on.... there's a non commercial G729 codec that will work with Asterisk? Can someone point me to where I can find it? Thanks, Doug. -----Original Message----- From: Francesco Peeters (Asterisk) [mailto:francesco@fampeeters.com] Sent: Sun 1/22/2006 8:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion