Thanks for the responses on this one
David Gibbons: reinvite=no is set, as we need the asterisk box to maintain the
audio for recording... (I believe even if we didn't have this option,
MixMonitor would have the same effect anyway.)
Peder: the firewall is integrated into the router, and is
On Wed, 2010-01-20 at 23:41 +0100, Michiel van Baak wrote:
Forget about virtualization!
This system is running linux as base os (I conclude by the tone of your
mail)
Just install asterisk on it besides the monitoring software and be done
with it.
What do you gain by running virtualisation on
Hi everybody,
I would like to use realtime authentification with my LDAP.
My Asterisk is v. 1.6.1.12. I'm using OpenLDAP
The command realtime ldap status is OK.
I have configure these files :
/etc/asterisk/extconfig
/etc/asterisk/res_ldap.conf
/etc/asterisk/extensions.ael
I do nothing and I
Hello,
I wrote a little AGI-Script that implements an IVR (using asterisk 1.6).
The whole conversation is recorded and at some points the caller should
tell some information.
I detect the silence (WaitForSilence) to go to the next step in the IVR.
Until now everything is OK, but...
some
You last question : why are DTMF tones not audible in the recording?
WE had issues with DTMF not recording, and found it was due to the handset only
sending the DTMF in data, rather than inline, as a beep... that could be your
reason :)
Steven Davison
Net Technial Solutions
-Original
We have been successfully using Asterisk (1.6.0.x) in a heavily loaded
Virtuozzo (= commercial OpenVZ) environment for over a year. I'm sure we
aren't the only ones to do so.
We had some terrible problems with random one-way audio a few minutes into
some calls to start with, which I was worried
Hi,
I'm having trouble getting feature codes to work in Asterisk 1.4.21.2.
Features.conf contians this:
blindxfer=##
atxfer=*2
automon=*1
disconnect=**
I'm really most interested in getting disconnect to work so that I hear
Goodbye when I press ** during a call connected this way in my dial
Hi,
I'm having trouble getting Dial to exit when the caller hangs up in Asterisk
1.4.21.2.
I use a POTS line to call into the DiD given to me by VOIP service
provider. When the call comes in, I have the VOIP provider send it to
another POTS line. All this works fine however when the caller
Since there are no DAHDI lines involved, polarity probably won't help.
Call-limit might or might not help with this. Does core show channels
show anything after the callee hangs up?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Hi,
Couple of questions...
Are you allowing reinvites, and what happens if you change the dialplan to this?
exten = 1,n,Dial(SIP/14168724...@6135551212-sw1|120|gtT)
exten = 1,n,Playback(vm-goodbye)
exten = 1,n,Hangup()
help this helps :)
Steven Davison
Net Technial Solutions
From:
Hello,
I want to know what is timeout for MS SQL connection? My config is:
[mydb]
enabled = yes
dsn = MYDB
pooling = yes
limit = 200
share_connections = no
username = login
password = password
pre-connect = yes
backslash_is_escape = no
In the peak , I can see :
ODBC DSN Settings
On Thursday 21 January 2010 09:51:13 Giedrius Augys wrote:
Is it possible to free idle connections? When limit was 40, I had lost part
of data. My asterisk version is 1.6.0.20 .
We intentionally do not, since the maximum number of connections is always
the maximum concurrent number of queries,
I am having problems with spa3102 FXO ports. It has a lot of echo, so be
careful when you get one of does!
2010/1/6 Joseph L. Casale jcas...@activenetwerx.com
I don't use them myself, but I was thinking that the RHEL5 spec files
might be another place to look for what you need to build with
Hello all,
I have a Linksys spa3102 with one FXS and one FXO port.
The problem is that I have a lot of echo when using the fxo port, the sound
is of very low quality.
So, since I am passing from a FXO port to a SIP channel I ask:
is there any Sip echo canceler software for
Hi, I´m trying to use SS/ in Asterisk.
I'm thinking in chan_ss7 and libss7, and I want to know some other
experience with this.
Thanks!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Problem solved. :)
I was adding to the pstn line a gain of 6 DB for both sides.
It has to be less than zero. After that the echo almost disappeared.
2010/1/21 Alexandre Rodrigues alex...@gmail.com
Hello all,
I have a Linksys spa3102 with one FXS and one FXO port.
The problem is that I
On Wed, 2010-01-20 at 17:06 -0800, Jim Dickenson wrote:
My development system for asterisk is a virtual CentOS 5.4 world running
under Fusion on my MacBook. I am usually only doing a few calls at a time. I
have an IAX trunk to our office Asterisk PBX so I can access the PRI line
there. I do
I am in the process of trying to terminate a PRI into a new * server. The
server has an old T100P T1/PRI card in it. I have compiled the following on
Centos 5.4.
dahdi-linux-complete-2.2.1+2.2.1
libpri-1.4.10.2
asterisk-1.4.29
Everything seems to have compiled fine. DAHDI reports Found a
This is often caused by the dahdi module not loading, check
/var/log/asterisk/messages for the reason, or better yet, from the cli load the
module manually and see the error in real time. If I had to guess I would say
it is a configuration error.
Thank you and have a nice day,
Anthony Francis
Thanks you were exactly right. I had a problem in my chan_dadhi.conf file.
Basically, I had the channels defined before the signaling and it wouldn't
load. It did not show any errors that I could see on startup and there were
no messages in the /var/log/asterisk/messages but when doing a load
In the good old days ( ZAPTEL ) Channel = had to be the last line in
a section. Everything after channel = was ignored.
Seems that has not changed.
A hard learned lesson by many !
Eric Merkel (Mail Lists) wrote:
Thanks you were exactly right. I had a problem in my chan_dadhi.conf
file.
Most manufacturers charge in excess of $80 to upgrade from a 10/100
switch to a 10/100/1000 switch built into the phone.
The cost might have been in the chipset 5 years ago but I can get a 5
port gigabit switch for $30.
What are most folks using for people that need gigabit to the desktop
and
-Original Message-
From: Felix Tiefenthaler [mailto:tiefenthale...@gmail.com]
Sent: Wednesday, January 20, 2010 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Virtual Asterisk Installation
Hi all!
I've been reading this list for a few weeks
Hi everybody,
I would like to use realtime authentification with my LDAP.
It depends on what you are doing with LDAP. There is an LDAP realtime engine
for SIP/IAX peers, voicemail users, asterisk configurations and extensions with
a sample ldif included with the distro, although I
On Thu, Jan 21, 2010 at 4:56 PM, Matt Darnell mattdarn...@gmail.com wrote:
Most manufacturers charge in excess of $80 to upgrade from a 10/100
switch to a 10/100/1000 switch built into the phone.
The cost might have been in the chipset 5 years ago but I can get a 5
port gigabit switch for $30.
Thanks responding guys. It appears that it's the canreinvite that's causing
the problem. Interesting results tho:
With canreinvite=yes, leaving out the transfer options leads to a Dial
command that _never_ exits:
exten = 1,n,Dial(SIP/14168724...@6135551212-sw1|120|g)
I have 2 channels
Hi,
I'm having trouble getting feature codes to work in Asterisk 1.4.21.2.
Features.conf contians this:
blindxfer=##
atxfer=*2
automon=*1
disconnect=**
I'm really most interested in getting disconnect to work so that I hear
Goodbye when I press ** during a call connected this way in my dial
At 9:08 PM on 21 Jan 2010, hugolivude wrote:
The call works fine and the CLI tells me that ** is an active feature:
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# ##
Attended
On Thu, Jan 21, 2010 at 3:30 PM, Jonathan Thurman jthurma...@gmail.com wrote:
On Thu, Jan 21, 2010 at 4:56 PM, Matt Darnell mattdarn...@gmail.com wrote:
Most manufacturers charge in excess of $80 to upgrade from a 10/100
switch to a 10/100/1000 switch built into the phone.
The cost might have
Thanks for the reply. I'm not convinced it's a DTMF problem anymore because
I tried all the options still no luck :-( Also I'm dialing the number from
an IVR menu so it is recognizing the '1' that i press from the menu. Any
other ideas I could try? I am supposed to put this:
include =
2010/1/21 Tilghman Lesher tles...@digium.com
On Thursday 21 January 2010 09:51:13 Giedrius Augys wrote:
Is it possible to free idle connections? When limit was 40, I had lost
part
of data. My asterisk version is 1.6.0.20 .
We intentionally do not, since the maximum number of connections
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