Displaying 20 results from an estimated 300 matches similar to: "Having problems with Agents and calls going to voicemail"
2018 Feb 08
0
Cannot delete IMAP Mail-Folder in Trash
> I have the same Problem with thunderbird on Linux.
>
> It looks like thunderbirds imap-implementation is a little bit broken.
> Even if thunderbird tried once to get a mailbox, it persists in the
> profile.
>
> I have FS-Layout in dovecot. Thunderbird somtimes try to get a
> submailbox with INBOX/firstlevel^secondlevel^thirdlevel
>
> Then it doesn't find this
2018 Feb 06
1
Cannot delete IMAP Mail-Folder in Trash
We have the same problem, with a twist. When Thunderbird deletes a folder, it is still shown by the GUI. Dovecot deleted the folder correctly, and the sunscriptions file is also correct. Some other times, on shared folders, Thunderbird refuses to delete; in this case, apple mail on iphone can delete successfully. This suggests that the problem is in Thunderbird's code.
R
On Mon, Feb 5, 2018
2007 Feb 01
1
Dial option G - Passing parameters?
Has anyone used the G option with the Dial app? I'm looking for a way
to control the called party leg. Specifically, I'd like to pass a few
variables to the called side for some call control. Here's a synopsis
of what I'm doing:
Make outbound call w/ AMI Originate action.
Called party answers ("Customer")
Customer identifies himself, and now I use Dial w/ the G
2003 Sep 07
2
New cvs compile; basic operational question, please.
Can someone offer a hint on what I'm doing wrong with the basic * config?
Just implemented * for the first time using yesterday's cvs. The initial
configs are based on John Todd's article at http://www.onlamp.com/lpt/a/3956,
and using two 7960's for initial testing. When one 7960 calls the other, I
get the following and the call is dropped:
-- Executing
2006 May 16
0
Need help with Dial M option and destination context
I would appreciate hearing from anyone who has figured this one out.
Here's the scenario:
I have a context wherein I give the called party the option to dial the
digit 9. If he does so, he is transferred a la this extension entry:
exten => 9,1,Playback(pls-hold-while-try)
exten => 9,n,Noop(Attempting to bridge to ${agentext})
exten =>
2005 Aug 25
0
Internal FXS to SIP problem
I've just setup a new asterisk box (cvs HEAD) with a digium tdm411 and
a couple computers with eyebeam. I have one small. I cannot call the
eyebeam clients from the phone connected the fxs port. I can call the
phone from the eyebeem clients. And, I get both the fxs phone and
eyebeam clients to ring when a call comes in through the fxo port.
I have been trying to get this straightened out
2006 Aug 11
2
AgentcallbackLogin()
Can someone tell me why this is not valid...
[start]
exten => 1000,1,Answer
exten => 1000,2,Wait,1
exten => 1000,3,AgentcallbackLogin(1000||2000@Local)
exten => 2000,1,Macro(DialProxy,115551212)
exten => 3000,1,Queue(testq||||45)
while this is:
[start]
exten => 1000,1,Answer
exten => 1000,2,Wait,1
exten => 1000,3,AgentcallbackLogin(1000||2000@start)
exten =>
2005 May 25
5
how to dial extension with menu
hello
like if 6000 is the main exchange number. any one dial
to 6000 will be asked for pressing his desired
extension then he can press his desired extension then
his number is diled
exten=>6000,1,Background(enterdesiredexten)
exten=>6000,2,Wait(2)
exten=>2000,1,Dial(SIP/${EXTEN})
2013 Sep 24
0
Problems starting vtmmgr
Hi,
I am following http://xenbits.xen.org/docs/unstable/misc/vtpm.txt, but
I''m having some problems when I try to start vtpmmgr-stubdom
I''m using Xen 4.3 on Ubuntu 12.04 and I have a physical TPM.
The config file for vTPM manager is:
kernel="/usr/local/lib/xen/boot/vtpmmgr-stubdom.gz"
memory=16
disk=["file:/var/vtpmmgr-stubdom.img,hda,w"]
2006 Nov 08
0
Warning: "Channel does not have a CDR" when doing ForkCDR
Gang,
I'm having this error pop up when I do a ForkCDR, and I'm not sure how
to get around it. Here are a few log lines:
Nov 8 10:37:08 VERBOSE[28079] logger.c: -- Executing
ForkCDR("Zap/49-1", "") in new stack
Nov 8 10:37:08 WARNING[28079] app_forkcdr.c: Channel does not have a
CDR
The scenario occurs like this:
I use a .call file to generate a call on
2017 Jun 18
2
R_using non linear regression with constraints
I am using nlsLM {minpack.lm} to find the values of parameters a and b of
function myfun which give the best fit for the data set, mydata.
mydata=data.frame(x=c(0,5,9,13,17,20),y = c(0,11,20,29,38,45))
myfun=function(a,b,r,t){
prd=a*b*(1-exp(-b*r*t))
return(prd)}
and using nlsLM
myfit=nlsLM(y~myfun(a,b,r=2,t=x),data=mydata,start=list(a=2000,b=0.05),
lower = c(1000,0),
2017 Jun 18
3
R_using non linear regression with constraints
https://cran.r-project.org/web/views/Optimization.html
(Cran's optimization task view -- as always, you should search before posting)
In general, nonlinear optimization with nonlinear constraints is hard,
and the strategy used here (multiplying by a*b < 1000) may not work --
it introduces a discontinuity into the objective function, so
gradient based methods may in particular be
2005 Jul 15
2
[Aserisk-Users]no audio inside the net
Hi list, i've problems with my * server and the 4 phones which are
linked to it. i've 2 grandstream bt100 with the firmware upgraded to
101, a wi-fi phone (i don't know its brand) and another ip phone i
don't know its brand. with this sip.conf :
[general]
port = 5060
bindaddr = 192.168.100.229
context = default ;x changed from default to sip
localnet = 192.168.100.0/24
2004 Jun 28
2
Would this work?
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)
2005 Mar 09
6
how to sip->h323 using asterisk-oh323-0.7.1
hello
i am using asterisk-oh323-0.7.1. i want to convert sip
call to h323 (h323 sjphone or h323 proxy). what could
be the best way for this. i am successfull in
converting h323->sip by using asterisk as gateway.
help required on sip->h323.
kamran
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2017 Jun 18
0
R_using non linear regression with constraints
I ran the following script. I satisfied the constraint by
making a*b a single parameter, which isn't always possible.
I also ran nlxb() from nlsr package, and this gives singular
values of the Jacobian. In the unconstrained case, the svs are
pretty awful, and I wouldn't trust the results as a model, though
the minimum is probably OK. The constrained result has a much
larger sum of squares.
2017 Jun 18
0
R_using non linear regression with constraints
> On Jun 18, 2017, at 6:24 AM, Manoranjan Muthusamy <ranjanmano167 at gmail.com> wrote:
>
> I am using nlsLM {minpack.lm} to find the values of parameters a and b of
> function myfun which give the best fit for the data set, mydata.
>
> mydata=data.frame(x=c(0,5,9,13,17,20),y = c(0,11,20,29,38,45))
>
> myfun=function(a,b,r,t){
> prd=a*b*(1-exp(-b*r*t))
>
2017 Jun 18
3
R_using non linear regression with constraints
I am not as expert as John, but I thought it worth pointing out that the
variable substitution technique gives up one set of constraints for
another (b=0 in this case). I also find that plots help me see what is
going on, so here is my reproducible example (note inclusion of library
calls for completeness). Note that NONE of the optimizers mentioned so far
appear to be finding the true best
2005 Feb 10
4
why asterisk is replying 404 Not Found
[3000]
type=friend
dtmfmode=INFO
insecure=yes
canreinvite=no
auth=plaintext
host=dynamic
allow=ulaw
[2000]
type=friend
dtmfmode=INFO
insecure=yes
canreinvite=no
auth=plaintext
host=dynamic
allow=ulaw
i have declared these two users 3000 and 2000. they
are registering successfully.
problem is that
2009 Oct 22
1
queues autopause
Hi,
I have 3 queue set in the table as below.
name,autopause
1000,1
2000,1
3000,1
In queue 1000, the autopause works after member failed to answer call.
However, other queues don't work for the autopause function.
queue 1000:
-- Nobody picked up in 25000 ms
-- Auto-Pausing Queue Member SIP/1234 in queue 1000 since they
failed to answer.
queue 2000/3000:
-- Nobody picked up in