Displaying 20 results from an estimated 2000 matches similar to: "urgent outbound dialing problem"
2004 Nov 21
4
UK available SIP phone?
Hi,
Anybody here from the UK using Asterisk at home?
I'm looking for a SIP phone which will work with Asterisk and
not leave me broke!
I got one of the Tecom ones from Solwise but it refuses to
login to Asterisk server for some reason. May have to send it back.
What are the other options please?
Thanks
Mike
2002 Apr 23
2
Asking about how to use R to draw Time Series graph
Hi
I'm study at University of Canterbury. Now, We have one project use R
to do time series. The problem is I don't know how to use R to draw time
series graph! Can you help me sovle this problem? I can not find in
manuals book! Can you tearch me what fuction command I have to use?
Thanks alway!
Sam
2005 Jan 16
2
[LLVMdev] LLVM 1.4 Build Error
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=GB18030" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
Hello,<br>
I built llvm1.4 with gcc3.4.3 and glibc2.3.2 on a redhat 9.0 machine,
2004 Dec 13
1
Asterisk up & running, now what?
Hi,
I've recently got * working (thanks Clive and list!) at home. We have
2 PSTN lines
connected via X100P cards. I've got 3 x SIP phones (2 are Budgetone,
the other is
a Tecom SIP).
One of the lines is our standard home line, the other a business line. Presently
I've got * set so you dial 91<tel no> and 92 <tel no> to select which
line to dial out on.
I should probably
2005 Jan 16
0
[LLVMdev] LLVM 1.4 Build Error
The HAVE_MMAP_FILE macro comes from when you configured llvm. The configure
tested if your system is giving access to a various resources required by
llvm. LLVM needs the MMAP functionality by your system. Some times, if the
required resources aren't present, we use other resources.
However, in this case the building process is stopped, because the
functionality can't be implemented
2010 Mar 11
4
help about solving two equations
I have two matrix s1 and s2, each of them is 1000*1.
and I have two equations:
digamma(p)-digamma(p+q)=s1,
digamma(q)-digamma(p+q)=s2,
and I want to sovle these two equations to get the value of x and y, which are also two 1000*1 matrices.
I write a program like this:
f <- function(x) {
p<- x[1]; q <- x[2];
((digamma(p)-digamma(p+q)-s1[2,]) )^2 +((digamma(q)-digamma(p+q)-s2[2,]) )^2
2009 Dec 17
2
Integrate a CPE with Asterisk in MGCP
Hello all,
I'm looking for some help to try to understand why my CPE doesn't work
good with Asterisk in MGCP.
Here is what I want to do :
- Register a TECOM AH4021 on Asterisk in MGCP with the following profile
in mgcp.Conf :
[general]
port = 2727
bindaddr = 10.95.20.1
disallow=all
allow=g729
allow=alaw
020202020202]
context=mgcp
host=dynamic
canreinvite=no
dtmfmode=rfc2833
nat=yes
2008 Jun 25
0
[LLVMdev] Using annotation attributes
Hi all,
I've also been developing an interest in using IR annotations for my compiler.
Some discussion with Bart turns out that he has implemented some code to parse
the llvm.globals.annotations array, but in no way integrated or reusable.
We've spent some thought about how this could be done properly, which I will
share here.
Firstly, however, I was wondering about the format of the
2004 Dec 01
1
conference room possible bug
hi;
i setup a Meetme conference room and i notice the following behavior:
if A calls confroom over PSTN channel 1
B call confroom over PSTN channel 2
C calls confroom over SIP/Ethernet
then i have all of them talking and the media stream mixed by asterisk.
However, if i hang up A, channel 1 is still ocuppied (i try dialing
inbound again on channel and it continues to give a busy siganl)
any
2004 Dec 02
2
threeway calling
any idea on how we can setup threeway calling in *
thanks
moe smadi
2005 Mar 11
1
digium card
hi;
does any body know what are the physical dimension of a digium care
400pm for example?
thanks
m.smadi
2004 Nov 21
0
sip debug command?
Hi,
Whilst trying to get this Tecom phone working with Asterisk, it seems
to be unable to login. Using the 'sip debug' command from the CLI does
not produce any
output even though the debug of the phone shows it trying to login every
second or so?
The phone seems to be based on a "Centrality PA1688" processor.
Is there any other way I can see why this phone fails to login?
I
2005 Aug 12
3
TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a
TDM400P with (1) FXO card on port 4. Inbound calls are always successful
but outbound calls fail 75% of the time with intercept messages from my
dial tone provider that include "we're sorry, your call did not go
through", and "we're sorry, when placing a local call it is now
necessary to dial an area
2005 Aug 31
1
Softphone vmail indicator and TDM400P woes
Hello list...
1) Is there an IAX softphone that supports any kind of voicemail indicator?
2) I have 2 TDM400Ps installed in a system. I need the audio on the
analog phone (FXS modules) to be amplified somewhere between 10 and
15db so I set rxgain and txgain to 15 for the FXO interfaces and FXS
interfaces. When a call comes in on the FXO at this setting, the call
sometimes has about 20 seconds of
2007 Feb 19
1
random effect nested within fixed effects (binomial lmer)
I have a large dataset where each Subject answered seven similar
Items, which are binary yes/no questions. So I've always used Subject
and Item random effects in my models, fit with lmer(), e.g.:
model<-lmer(Response~Race+Gender+...+(1|Subject_ID)+(1|
Item_ID),data,binomial)
But I recently realized something. Most of the variables that I've
tested as fixed effects are properties
2008 Oct 23
0
command - set sip_codec- does not work with asterisk-1.4.21
hello:
i want to test the g729 with asterisk. my scenario is sipp(ulaw)->asterisk1 with g729->asterisk2 with g729.
I want to test g729 module with asterisk-1.4.21, when i make calls from asterisk 1 to asterisk 2, the asterisk 1 always send ulaw to asterisk 2. my sip in asterisk 1 is with codec g729 and enforce that use g729, the sip in asterisk 2 also work with G729 only, but asterisk 2
2012 Jun 23
0
[LLVMdev] Why can not sparcv9 backend handle i64 produced by FrameIndex?
Hi, all,
I have been recently porting a backend for our experimental DSP.
It has a regular register file for ALU, naming it R registers, and
another register file (J registers) for memory access.
Both R registers and J registers are 32-bit.
Since LLVM cannot distinguish 32-bit integers or pointers during
register allocation, I have to define J as 64-bit, although
it's physically 32-bit. This
2004 Dec 15
2
TDM400p FXO module always offhook
I have a TDM400p with 3 FXS mods and 1 FXO mod. I have all set up with
what seems to be correct settings (according to digium and asterisk wiki).
As soon as I plug in my POTS line into FXO mod the line goes into offhook
state (whether I have power to the card or not). Should this happen?
When I try to call * box all I get is busy signal. I've installed stable
version, cvs version, change
2008 Jun 05
5
[LLVMdev] Using annotation attributes
Hi,
I'm trying to annotate certain functions in C code, and do something with
these functions in my LLVM pass. I annotate the C code like this:
int __attribute__((annotate("annot"))) function() {
This nicely gets added to the LLVM bitcode in an
@llvm.global.annotations global. Now I had hoped that it'd be easy to extract
a list of functions annotated with my annotation
2004 Dec 15
3
Newbie setup (Hardware questions)
Hello, I'm trying to setup an Asterix PBX solution in
our office.
We plan to have 5 active lines open available at any
point in time.
We'd like to use VoIP Phones, and possibly Software
Based phone (*NIX/Windows enviroment).
I was researching the various cards and I think I'd
want to go with the Digium TDM40B - 4-port.
However, I can't figure the differences between FXS &