Displaying 20 results from an estimated 300 matches similar to: "Creating extension groups"
2005 Feb 23
3
Send outgoing calls to a SIP gateway
How do I route all the outgoing calls through a SIP gateway, this should send more than one outgoing call to the sip gateway at once. please show me the sample configurations on how to do this.
my SIP gatway can accecpt direct IP traffic or SIP proxy traffc.
Thank You
Kanishka
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2005 Sep 29
4
OOH323C
hi
has any one used OOH323C i tried this it is installed but do not know how to
configure has any one used this, what is the best h323 addon to use with
asterisk
2005 Feb 27
1
limit SIP extention outgoing calls
Hi
how do i set an SIP users to make outgoing calls that is worth only $5. if they exceed $5 they can't make any calls. what i need is not a calling card, but to limit outgoing calls for SIP users depedning on a value i give.
I use realtime asterisk.
Thank You
Kanishka
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2006 May 04
2
Asterisk on amd SERVER
Hi
I am going to install asterisk on an AMD server, did any one had problems
installing it on an AMD server ?
Regards
Kani
2005 Jan 23
3
Asterisk 1.0.5
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hello everyone,
As you know, we released Asterisk 1.0.4 earlier this week. However,
there was a callerid bug in chan_zap that has caused us to go ahead and
make another release. Asterisk 1.0.5 is available at all of the usual
locations.
I'm sorry for any inconvenience this may cause.
Russell Bryant
-----BEGIN PGP SIGNATURE-----
Version: GnuPG
2005 Mar 15
1
oh323 and open 729
has any one installed this, i just tried this on a test server, i get voice but it's corrupted, i do not get the natural voice
any idea why
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2005 Feb 25
1
Asterisk and 723,729
has any one implemented asterisk with 723 and 729 codecs, what is the cheapest way.
is there a limitation in the open 723 implementation ??
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2005 Sep 28
15
Asterisk on windows
why can't we compile the asterisk coading in windows, it's done in c++ so it
should work in windows as well
2005 Feb 24
2
Polycom Call Parking
Has anyone gotten the call parking soft button on the polycom
phones(specifically the IP 600) to work with call parking that asterisk
provides? If so what configuration changes were needed?
--johann
2014 Jun 26
2
[LLVMdev] problem with X86's AVX assembler?
On Thu, Jun 26, 2014 at 10:23 AM, Adam Nemet <anemet at apple.com> wrote:
>
>
> On Jun 25, 2014, at 7:05 PM, Jun Koi <junkoi2004 at gmail.com> wrote:
>
>
>
>
> On Thu, Jun 26, 2014 at 5:47 AM, Adam Nemet <anemet at apple.com> wrote:
>
>> Hi Jun,
>>
>> On Jun 25, 2014, at 8:14 AM, Jun Koi <junkoi2004 at gmail.com> wrote:
>>
2014 Jun 26
2
[LLVMdev] problem with X86's AVX assembler?
On Thu, Jun 26, 2014 at 5:47 AM, Adam Nemet <anemet at apple.com> wrote:
> Hi Jun,
>
> On Jun 25, 2014, at 8:14 AM, Jun Koi <junkoi2004 at gmail.com> wrote:
>
> > Hi,
> >
> > I am trying to assemble below instruction with latest LLVM code, but
> fail. Am I doing something wrong, or is this a bug?
> >
> >
> > $ echo "vaddps zmm7
2014 Jun 25
2
[LLVMdev] problem with X86's AVX assembler?
Hi,
I am trying to assemble below instruction with latest LLVM code, but fail.
Am I doing something wrong, or is this a bug?
$ echo "vaddps zmm7 {k6}, zmm2, zmm4,
{rd-sae}"|./Release+Asserts/bin/llvm-mc -assemble -triple=x86_64 -mcpu=knl
-show-encoding -x86-asm-syntax=intel
.text
<stdin>:1:31: error: unknown token in expression
vaddps zmm7 {k6}, zmm2, zmm4, {rd-sae}
2005 Mar 24
0
R: music on hold error
I've got the same problem. MusicOnHold works if I use something like:
Exten => 1111,1,MusicOnHold()
but if I try to answer a call and then transfer or put on hold the call, I get no music.
Does anyone have any idea?
Bye,
Gianluca.
_____
Da: Kanishka Somaratne [mailto:kani@technoportal.biz]
Inviato: gioved? 17 marzo 2005 5.53
A: asterisk-users@lists.digium.com
2005 Sep 30
0
oh323 implementation 0.67 has call-id problem
I am trying oh323(version 0.67) , make call from sip UA to h323 gateway,
can't get Call-id pass from sip UA to h323 gateway, h323 always gets
call-ID sent from Asterisk as *. are there any configure to pass
the correct call-id from sip UA to h323 gateway? or this is a bug in
oh323 0.67?
how about oh323 0.73 ?
Mario
On 9/29/05, Kanishka Somaratne <kani@technoportal.biz>
2003 Oct 13
6
Asterisk Manager
Hello all,
Can I execute linux command like(ls, mkdir) through the Manager interface?
I can't seem to access the manual at digium.com. I keep getting 'Forbidden'
error. Looks like they are upgrading or something.
CF
2006 Dec 20
1
Incoming Lines Confusion
First off, please, for the love of God, don't cremate me, if I should
already know the answer to this!
I've installed a small setup for an office who wanted to be able to talk to
each other instead of having to rely on MSN to communicate. Weird request, I
know, but hey, we do what we need to do to get paid. I installed soft
phones, gave everyone an extension, and bingo, they can call and
2016 Oct 26
2
borrar texto en una gráfica
Hola a todos,
Os envío una consulta que considero sencilla pero me está resultando imposible de resolver. Si ejecutáis el siguiente código, obtendréis la gráfica que os adjunto:
library(ltm)
modelo <- rasch(LSAT)
plot(modelo, main="Curva probabilidad pregunta 1",legend = TRUE, cx = "bottomright", items=1,xlab="Conocimiento",ylab="Probabilidad")
Resulta
2003 Dec 30
2
E100P configuration
Hi !
I am trying to configure two E100P cards, but I am a bit confused with
zapta.conf in what I am trying to achieve.
The * will be connected to a pstn switch with two E1 PRI lines. The E1 lines
will be used for incoming calls as well as outgoing calls.
My problem now is what to put in zapta.conf, I would like to group all
channels from both cards together (if that's possible). Does this
2005 Mar 22
0
RE: Asterisk-Users Digest, Vol 8, Issue 150
The update worked like a charm! Hold music is as cheesy as ever!
Thanks much, this list is a life saver!
Dan
------------------------------
Message: 2
Date: Fri, 18 Mar 2005 09:16:59 -0600
From: Eric Wieling <eric@fnords.org>
Subject: Re: [Asterisk-Users] Redhat 9 Music on hold
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
2004 Sep 26
2
spandsp with TDM fxo card?
Has anyone made spandsp to work with a digium tdm fxo card?
I finally got the rxfax and txfax modules to compile, the spandsp lib
installed (and in the libpath), and now receive:
-- Starting simple switch on 'Zap/1-1'
-- Executing RxFAX("Zap/1-1", "/var/fax.tif") in new stack
-- Hungup 'Zap/1-1'
I've tried to adjust rxgain/txgain in