Displaying 20 results from an estimated 600 matches similar to: "Getting "ghost" transfer or music on hold"
2010 Nov 27
2
Preserve CallerID on transfers
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<font face="Arial">Hi, it´s possible to mantain the original
2009 Aug 01
1
how to setup incoming calls not to use authentication
Dear all,
In Sip.conf file how to setup incoming calls not to use
authentication?
Please provide some steps to do it..
Thanks...
Regards,
Velusamy
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2009 Oct 30
2
asterisk 1.6 enable cdr_mysql
How to enable cdr_mysql.conf in Asterisk 1.6?
I have installed asterisk-addons which compiled mysql support,
"module show" is showing "cdr_addon_mysql.so"
but cdr_mysql.conf was not created in /asterisk directory
Is there any configuration file to enable mysql support?
Comping cdr_mysql.conf from previous installation does not do anything, calls aren't recorded.
--
2010 Mar 11
2
Codec preference
How can I set the prefered codec between 2 calling parties ??
My Grandstream supports G729, alaw and gsm... in this order.
The Zoiper softphone has alaw and gsm as codecs... in that order.
Although there should be a matching codec found, my Grandstream can not
call the Zoiper softphone.
CLI shows :
[Mar 11 17:47:21] WARNING[22367]: channel.c:3340
ast_channel_make_compatible: No path to
2009 Dec 18
2
To Asterisk AMI Gurus - Tacking issue with originate
Hello Everyone,
I am making a simple index.php file which will allow a web user to enter his
$phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged.
Following is the index.php and the contents of extensions_custom.conf. When
I submit the form nothing happens. I don't even see Manager Connected msg.
Your input will be much appreciated. I am thinking I have some syntax
2004 Sep 10
1
(Resend) Trouble with all linux sip softphones.... And asterisk/linphone/kphone SRPMs
Got no responses to this, but the list seemed to be down for a while, so
here it is again. Sorry for the extra bandwidth!
John
Hi, I've been messing with getting SIP working for days now, with
limited success. I've got Asterisk set up on a remote server with the
echo test. Please try it out to verify I've got the server working
right:
sip:robot at nixon.butchwax.com
2004 Aug 18
1
Small patch to zaptel Makefile
Minor fix. I'm using this in my RPM specfile.
John
--- ./zaptel-1.0-RC1/Makefile.bigu 2004-07-16 17:09:07.000000000 -0500
+++ ./zaptel-1.0-RC1/Makefile 2004-08-18 16:28:45.000000000 -0500
@@@ -316,10 +318,10 @@
elif [ -d $(INSTALL_PREFIX)/etc/init.d ]; then \
install -m 755 zaptel.init
$(INSTALL_PREFIX)/etc/init.d/zaptel;\
\
fi
- if [ !
2005 Mar 13
2
How can I eveluate trailing numbers in extensions.conf?
Checkout
http://www.voip-info.org/wiki-Asterisk+variables
I believe that should have the answer for you.
furthermore assuming that your number is always going to be 12 digits.
exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number.
Hope this helps.
Umar
On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote:
> Hi,
>
> this
2010 Sep 07
1
Solving the CDR mess of attended transfers
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="Arial">Is there a way to solve the mess on CDR caused by
CDR
2005 May 16
1
Problem with LTSP and WINE
Hi guys, (sorry for my english, i?m brazilian)
I?m making a solution for my business , using LTSP and WINE for run my
application for Linux...
I installed sucessful both ltsp and wine, i?m running perfectly my
application on server of LTSP...
but when i try to run the same application on the terminal?s of LTSP, i
receive the following error:
X Error of failed request: BadShmSeg (invalid
2010 Mar 23
4
Safe_asterisk doesn't exists???
Hello my friends,
I'm very worry about a problem i'm having...my asterisk got freez some
times, every 5 or 6 days with NO trace in /var/log/asterisk/messages
What i want to know is if safe_asterisk has something to be with this?
This is what i have on my server:
[root at mypbx ~]# ps -A | grep asterisk
9118 ? 00:01:30 asterisk
[root at dreampbx ~]# ps aux | grep asterisk
root
2009 Dec 30
1
problem with ring being sent to caller
I am using asterisk 1.6.0 and -- not all the time -- when a caller comes
in and my ivrdials an extension, the ring he gets sounds like a modem
handshake instead of the normal ring tone and it only sounds once even
if the phone is not picked up. Anyone seeing this -- the logs look fine
as far as I can tell.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you
2013 Aug 28
1
named lmer.models in do.call(anova,models)
Hi,
For some reason do.call on anova fails if the models are named lmer objects.
Consider the following example:
library(lme4)
models <- list(
lmer(Reaction ~ Days + (1| Subject), sleepstudy),
lmer(Reaction ~ Days + (Days | Subject), sleepstudy))
#
# models is an unnamed list, do.call works (although with warning):
do.call(anova, models)
#
# after labeling the models, do.call gives an
2010 Apr 13
1
Interesting One Way Audio
I have an Asterisk box, 1.4.30 with a PRI.
A Mitel 3300 is connected to the Asterisk box via SIP trunking.
When a user calls from the Mitel through the Asterisk box the user can speak
but can not hear the far end.
But - when I route the Mitel user to echo() it works, send and receive. The
Mitel user also can record and playback greetings.
One thing I have noticed is that when the Mitel user
2004 Dec 22
1
MGCP Transaction identifiers
I know this is not the most appropriated list to this, but I will try:
Does anyone know what is the criteria to the generation of the transaction identifiers in MGCP? I mean, are they generated by a randomic method?
I'm using Asterisk and MGCP eyeP Phone and observed that the RSIP and NTFY (methods created by the gateway) use high values in the transaction identifiers, while the RQNT, AUEP,
2009 Dec 28
2
Registering with a static peer?
I've been using a couple of Polycom 501 phones in my home Asterisk setup. I set
up each phone in sip.conf to be static, i.e. host=<phone ip address> so that
registration wasn't required. This has worked fine for me for a couple of years.
Now I just bought a Polycom 335. Since the 501's are now obsolete, I had to go
through the steps required in order to have separate
2004 Aug 24
0
Trouble with all linux sip softphones.... And asterisk/linphone/kphone SRPMs
Hi, I've been messing with getting SIP working for days now, with
limited success. I've got Asterisk set up on a remote server with the
echo test. Please try it out to verify I've got the server working
right:
sip:robot@nixon.butchwax.com
Running FC1, ThinkPad T22, headset thru the soundcard. Asterisk is
asterisk-1.0_RC1. No NAT. The phones I've tried so far are as
2005 Aug 07
0
Ghost Train automatic functional browser testing (pre-alpha)
Hi all,
As a proof-of-concept that recording and replaying events works to an
extend that it actually becomes useable, I''ve put up a proof-of-
conept technical demonstration here:
http://script.aculo.us/playground/ghosttrain/test.html
Note: It uses pure DOM functions, so it works in _Firefox_ only (1.0
+). (Safari lacks support for some of the DOM functions uses, and
Internet
2004 Mar 15
0
smbd/service.c:set_current_service(56) chdir (/var/ghost) failed
Hey Gang,
I have a suse 8.1 server that I have compiled samba 3.0.2 from source on.
I have it configured to authenticate against our AD. That seems to be
working fine.
getent passwd produces the results from both passwd and AD. getent group
gives me the groups + the AD groups. I was able to assign the permissions
to the folder without problems. It picked up the group from AD and
assigned it
2006 Jan 10
1
How to use rsync for Ghost or Acronis type backup
Madison,
That sounds like a really nice backup program. I'd like to try your
Beta out if I could. I've use rsync regularly and it would be great
to have a front end.
Thanks.
Thom