Displaying 20 results from an estimated 100 matches similar to: "No sounds on Playback()"
2005 Sep 21
1
I got "403", "Forbidden"... please help
Hi,
I'm setting up Asterisk as a voicemail with SER. My problem is,
when a caller that is not registered with asterisk (no username and
password in sip.conf) it prompts "403, Forbidden" . I need all calls
from outside of my network to reach asterisk for my users' voicemails,
because anonymous users will surely reach voicemail of my users to leave
messages. What do I
2005 Sep 21
4
How to retrieve voicemail from an IP phone?
Hi,
How can I retrieve those voicemails using my ip phone? and how
will i confiugre it on asterisk?
Please help I'm very new in asterisk.
Thanks,
--
Ryan Pagquil
Infodyne Inc. - PhilOnline.com
3603 Antel Global Corporate Center
Do?a Julia Vargas Ave.
Ortigas Center Pasig City
Tel: 687-0715
Web: www.philonline.com
2006 Jan 31
4
Asterisk Registering with SER question
Hi,
I've been registering asterisk to ser. I'm using SER as the outbound
SIP trunk for Asterisk. Users registered with Asterisk will use the
SIP trunk to reach SER registered users and PSTN's. Now when I
register Asterisk with SER, on my SER's location table I see these record:
Username Column = asterisk
Contact Column = sip:s@202.84.24.47
I have a script running that checks
2006 Mar 22
3
Voicemail limit?
Hi,
Is there an account limit for voicemail? I have 80+ users in the
voicemail and I can only reach the 70-ieth user. If there is a limit
how can I increase it to hundred for example?
Thanks,
Ryan
2011 Aug 23
3
Different Estimated values between R and Excel
Hi,
I used simple linear regression with the R software and EXCEL on the same
data. Although , I find the same R2=0.84, I find different estimated values
(intercept and slope). For the R software (slope =0.0009, Intercept =
-0.1478), for EXCEL (slope =927.7, Intercept = 154,41).
When I use the estimated values from the R software, the results seem bad,
however the results of Exel seem correct.
2005 Sep 23
1
Play sound on connect
Hello
A calls B, on connect I want B's greeting to be played to caller A.
I can see it is possible to play a sound to B on connect (DIAL(SIP/123
,A(hello)), but I cant se how to play a sound to A, is this possible?
Thank you
Michael
2006 Oct 22
0
[705] trunk/wxruby2/rake/rakedocs.rb: Add rake task to publish docs to wxruby website (Alex Fenton)
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.1//EN"
"http://www.w3.org/TR/xhtml11/DTD/xhtml11.dtd">
<html xmlns="http://www.w3.org/1999/xhtml">
<head><meta http-equiv="content-type" content="text/html; charset=utf-8" /><style type="text/css"><!--
#msg dl { border: 1px #006 solid; background: #369; padding:
2004 Dec 29
0
12 CANCEL's followed by 12 INVITE's in 5 secs
Hello All,
I have a problem that is alien to me and obvious for some of you
:). I have asterisk setup with few sip clients(using linphonec).
In a proper context, I have mentioned extensions 107 as
simputer@X.X.X.X (x.x.x.x=asterisk server ip)
Asterisk Sever-------------------------simputer(sip ua)
I can make calls from sipua to asterisk but not reverse way.
I get the following display on
2004 Dec 27
0
Call Placing timeouts
Hello All,
I have a problem that is alien to me and obvious for some of you
:). I have asterisk setup with few sip clients.
In a proper context, I have mentioned extensions 107 as
simputer@bogus.com
Asterisk Server-------------------------simputer(sip ua)
I can make calls from sipua to asterisk but not reverse way.
I get the following display on asterisk terminal
---------------------
2020 Jul 21
0
[PATCH v9 70/84] KVM: introspection: add KVMI_VCPU_GET_XSAVE
From: Mihai Don?u <mdontu at bitdefender.com>
This vCPU command is used to get the XSAVE area.
Signed-off-by: Mihai Don?u <mdontu at bitdefender.com>
Co-developed-by: Adalbert Laz?r <alazar at bitdefender.com>
Signed-off-by: Adalbert Laz?r <alazar at bitdefender.com>
---
Documentation/virt/kvm/kvmi.rst | 31 +++++++++++++++++++
2006 Jun 22
1
Re: Can I enter an extension to dial whilevoicemail is playing?
The options are not seperated by commas.
exten => s,1,Dial(SIP/50,23,r,d)
should be
exten => s,1,Dial(SIP/50,23,rd)
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of John Klimek
Sent: Thursday, June 22, 2006 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
2007 Aug 02
1
A simple IVR extension problem
Hi list,
I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS
5.
I am having trouble to make my simple IVR extension work, here is relevant
config:
zapata.conf
----
context=incoming
signalling=fxs_ks
channel => 4
context=internal
signalling=fxo_ks
channel => 1
-----
extensions.conf:
----
[office]
exten => s,1,Dial(Zap/1,30)
[home]
exten =>
2012 Nov 22
1
prediction problem
Hello,
I am using the mda package and in particular the fda routine to
classify/predict in terms of color to a set of 20 samples for which i don?t
know the color.
I preformed a flexible discriminant analysis (FDA) using a set of 147
samples for which i know all the information.
My script and data follow in attachment.
A total of 23 predictors were considered. 20 of the predictors are
2006 Jun 19
1
Can I enter an extension to dial while voicemail is playing?
I have a very, very simple Asterisk setup in my house. I have a
Sipura 3000 with a PSTN line connected and one analog phone connected.
The [incoming] context looks like this:
exten => s,1,Dial(SIP/50,23,r)
exten => s,2,VoiceMail(u50@default)
exten => s,3,Playback(vm-goodbye)
exten => s,4,Hangup
As you can see, when somebody calls in if I don't answer in 23 seconds
then they are
2006 Oct 18
3
Nightly automatic documentation generation
Is there a way to have RubyForge run our ''rake html_docs'' on SVN Head
each night and if successful copy to http://wxruby.rubyforge.org/doc/?
I do not know all the features RubyForge provides to us but this one
would be pretty nice.
Sean
2007 Jan 31
0
Random Sampling pointers?
Hello all,
I have a population of 112 servers that are experiencing different levels of packet loss. I don't want to poll all 112 of them (the analytical tools must be manually run on each individually) so it seems best to sample among them; then I plan on using R to run comparisons of the data pulled from each one. I'm not clear on the most sound way to go about this and I don't
2004 Dec 03
2
DIALSTATUS weirdness (CHANUNAVAIL instead of BUSY, NOANSWER instead of CHANUNAVAIL)
Just throwing this out here, hopefully someone can tell me why.
*CLI> show version
Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running
Linux
Zap/g1 is pri_cpe to Bell Canada
5551234 is a normal POTS line I have busied out (handset offhook)
exten => 1234,1,Dial(Zap/g1/5551234,,g)
exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
2011 Mar 11
1
[PATH 9/12] VTPM mini-os: New stubdom applications
This patch ports 5 new applications to the stubdom makefile structure
for inclusion into stubdom domains. While these are required for
vtpm-stubdom and vtpmmgrdom they could be used with other stubdom
applications.
-libgmp 4.3.2
-openssl 1.0.0a
-polarssl 0.12.1
-berlios tpm_emulator 0.6.1
-vtpm_manager (from the tools directory)
Signed off by: Matthew Fioravante
2016 Oct 27
0
[PATCH v2 5/7] nvc0: refactor TIC uploads to allow different specifics per generation
On Thu, Oct 27, 2016 at 1:19 PM, Samuel Pitoiset
<samuel.pitoiset at gmail.com> wrote:
> Are you sure this refactoring doesn't break anything?
>
> Few comments inline.
>
>
> On 10/27/2016 04:02 PM, Ilia Mirkin wrote:
>>
>> This flips GM10x to using the updated format, which is what I tested
>> with. However GM20x and GP10x also use this TIC format.
2016 Oct 16
0
[PATCH 4/5] nvc0: refactor TIC uploads to allow different specifies per generation
This flips GM10x to using the updated format, which is what I tested
with. However GM20x and GP10x also use this TIC format.
Signed-off-by: Ilia Mirkin <imirkin at alum.mit.edu>
---
src/nvc0_accel.c | 11 ++++++++++
src/nvc0_accel.h | 56 ++++++++++++++++++++++++++++++++++++++++++++++
src/nvc0_exa.c | 22 ++++---------------
src/nvc0_xv.c | 67