similar to: I can't use the 'Group', 'CallGroup' , 'PickupGroup' in SIP channel (asterisk1.4.2)

I''m having a problem formatting some text. I have a textarea that I input my text into, which will then be put into an e-mail and sent out. The problem I am having is when there is a single line break. Multiple line breaks work fine (hitting enter twice), but single line breaks don''t seem to work properly. Take the following input: Line1. Line2. Line3. Line4. Using

Here's a bit of a strange one. I hunted through the ocfs-users archives, but didn't see any mention of this, so here goes... Using coreutils 4.5.3-27 on RHAS 3, the utilities perform as expected against non-OCFS volumes: $ rpm -q coreutils coreutils-4.5.3-27 # cat a 3 line test file. Works as expected $ cat /var/tmp/file line1 line2 line3 # copy that file. Works as expected $ cp

Dear All, I am writing to ask a clarification on what R, and in particular lme, is doing. I have an experiment where fly wing area was measured in 4 selection lines, measured at 18 and 25 degrees. I am using a lme model because I have three replicated per line (coded 1:12 so I need not use getGroups to creat an orederd factor). The lines are called: "18"; "25";

I am trying to implement a rollover of extensions. exten => 3000,1,GotoIf($[${line1} = Congestion]?3:2) exten => 3000,2,Dial(${line1},15,rt) exten => 3000,3,GotoIf($[${line2} = Congestion]?5:4) exten => 3000,4,Dial(${line2},15,rt) exten => 3000,5,GotoIf($[${line3} = Congestion]?7:6) exten => 3000,6,Dial(${line3},15,rt) exten => 3000,7,GotoIf($[${line4} = Congestion]?1:8)

Hello, Let's say we have the following function: foo <- function(x) { line1 <- x line2 <- 0 line3 <- line1 + line2 return(line3) } And that we want to change the second line to be: line2 <- 2 How would you do that? The two ways I know of are either to use fix(foo) And change the function. Or to just write the function again. Is there

I currently have 4 lines coming into the house. We currently have an Avaya standard analog key system which has served us well, but running extensions is a major pain and requires a dedicated run per extension. I have ethernet run throughout the house though. The first two lines are "home" lines, however line1 I use for my consulting during the day, and line2 is used by my wife for

Dear friends, I met a problem on plotting. My dataset is : year MHBC LHBC MHRC LURC 1993 11.75 4.50 0.43 0.46 1994 7.25 1.25 0.35 0.51 1995 8.67 2.17 0.54 0.44 1996 2.67 1.33 0.78 0.47 1997 3.42 4.92 0.69 0.48 1998 1.92 3.08 0.72 0.54 1999 2.33 2.58 0.74 0.41 2000 5.75 4.50 0.45 0.50 2001 3.75 4.42 0.52 0.47 2002 2.33

Hi All; Asterisk is IP PBX, so there should be special IP Phones to be used with this IP PBX and this IP Phone should has special buttons working with Asterisk IP PBX, like: Line1, Line2, Line3, ... Line8 (these to be able to capture direct PSTN lines), also a button for Hold and transferring for another extension ... So can some one advise me which is the recommended IP Phone to be used with

Hi, I asked a time ago what were callgroup and pickup group used for. I have done some proofs and all, and I'm not sure if I have pick the idea up well!! That's what I understand: For example: group=1 callgroup =2 and pickupgroup=2 and my phone is a membership of the group 1. that's mean that when a phone that belong to group 2 is ringing, I'll be able to answer this call dialing

I have a file in the following format: Scenario1 Scenario1CIL Scenario1CIU Scenario2 Scenario2CIL Scenario2CIU 60 57 62 45 48 50 110 101 111 51 50 52 120 117 122 64 62 66 192 190 194 79 75 79 where: First column = Scenario1 mean value Second column = Scenario1 Low Confidence Interval Third column = Scenario1 Upper Confidence Interval Fourth column = Scenario2 mean value Fifth column =

I'm trying to set up asterisk so any phone connected to channel 1-16 of my Adit600 channel bank can pick up a call coming in on channel 24. I do not wish to ring any of the 16 channels on an incoming call -- this is strictly so I can pick up the line if I see it ringing and wish to answer at work. I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3. However

I have a 5-row matrix called “data”. There are headers. it look like this : Row 1 Row2 Row3 Row4 Row5 Line1 … … … … Line2 … … … … Line3 … … … … … Line 1838 … … … … I want to calculate several simple arithmetic means for Row5, that

Hi All. I've been experimenting with SLA on Asterisk 1.4.13 (patched up to 1.4.14). I am using a SIP channel for my "trunk" line. On the whole things are good, but I have noticed that if I misdial an outgoing call, i.e. I get 404 "Not Found" in the SIP trace, then the trunk line just drops, rather than presenting an error tone or message to the user.

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

Hi All, I am currently researching on doing high availability and load balancing with a couple of Linux servers, most specifically doing service takeover on top of Heartbeat. The scenario is: - 2 DSL lines from a single provider - 1 Broadband line from another provider Internet / | \ | | | /

Dear R users, Here is the coxme output I obtain on my survival dataset having 3 strains and 2 infection status (i: infected, ni: non infected) coxme(Surv(lay) ~ infection*strain, data=datalay, random= ~1 |block) Cox mixed-effects model fit by maximum likelihood Data: datalay n= 1194 Iterations= 3 77 NULL Integrated Penalized Log-likelihood -7270.028 -7223.859 -7218.175

I have problems using the call pickup under Asterisk 1.8.4.2. I have another Asterisk with 1.6 - and it is working fine with the same settings. I have setup the same callgroup and pickupgroup for all extensions in sip.conf - just to make things simple for testing. The sequence *8 seems to be completely ignored by Asterisk - the client shows "Call answered" when dialing *8 while the

I am trying to convert my 7905G to be SIP based and seem to be running into a few hassles. Below are all the config files and logs from the server. I have tried to follow the pdf's from cisco and some posts from other mailing lists that google turnedup, but it seems that nothing is working. Am I somehow missing a fundamental step in trying to upgrade from Call Manager to SIP? Any help is

I figured I''d post here before submitting a ticket, but I''m seeing some confusing stuff when dealing with YAML now. I was using it to freeze objects in my database, and so i had some data already around to mess with. I upgraded both Ruby (1.8.3 -> 1.8.4) and Rails ( -> 1.1RC1) and this junk started. I''ve outlined the problem in two pastes, which I''ll

> On Jan 2, 2024, at 23:17, thelma at sys-concept.com wrote: > > On 1/2/24 15:13, asterisk at phreaknet.org wrote: >>> On 1/2/2024 4:03 PM, thelma at sys-concept.com wrote: >>> I'm using asterisk-16.30.1 >>> >>> When I try to call another asterisk server over IAX I get a busy signal, >>> >>> chan_iax2.c:4739 __auto_congest: