Hello List,
I have one TOR3-E (E1 version) card from Govarion that i used some years
ago, but it seems company stopped work.
Since website is down.
Is there somebody with good heart that could help me to get a driver for an
x86 and x86_64 for
this card?
Thank you so much,
Richard Palmeron
--
Hi,
I am currently trying to get a Cisco Callmanager 4.1 and an Asterisk server
(1.6.2.21) to talk via a SIP trunk so I can use the Voicemail component of the
Asterisk (all the phones are associated with the Callmanager).
The connection seem to be there. When I do a sip show peers on the
Hello,
is there anyone that can give me some more information on these
deadlocks ?!
How can these deadlocks occur and what is good practise to avoid these
problems ??
Jonas.
Original Message
Subject:Re: [asterisk-users] Asterisk CLI unresponsive
Date: Fri,
On 6 February 2012 10:45, Jonas Kellens jonas.kell...@telenet.be wrote:
**
Hello,
is there anyone that can give me some more information on these
deadlocks ?!
How can these deadlocks occur and what is good practise to avoid these
problems ??
Jonas.
The only way to avoid deadlocks is
On 02/06/2012 12:14 PM, Steve Davies wrote:
On 6 February 2012 10:45, Jonas Kellens jonas.kell...@telenet.be
mailto:jonas.kell...@telenet.be wrote:
Hello,
is there anyone that can give me some more information on these
deadlocks ?!
How can these deadlocks occur and what is
Your running into a bug and the only way to solve it is to report it and debug
it and hope for a fix
There is no way someone can help without it being debugged and knowing what's
causing it to lockup
The only key to unlcock it when it gets locked is by restarting asterisk
Regards
When I compile Asterisk 10, under PBX modules I see DUNDi listed under extended
support. Does this mean that DUNDi will be depreciated in the upcoming
branches and if so what is replacing DUNDi?
Regards
David Klaverstyn
--
Why doesn't this manager.conf code work on Asterisk 1.6.2 and 1.8.9? It
works perfectly on Asterisk 1.4
In Asterisk 1.6 it appears to disconnect as soon as events occur and in
1.8.9 it can't be read at all. Apparently it has some syntax issues with
1.8.9.
Is it possible to tell at a glance what
When I do a make menuselect cdr_adaptive_odbc is disabled.
What packages are required to enable it?
Even after executing apt-get install unixodbc libmyodbc odbc-postgresql
tdsodbc unixodbc-bin it is still disabled.
What am I missing?
/voipfc
--
Hello,
I have a weird case, when some numbers dialed using a PRI, have an early
media sounds instead of normal ringing.
Few of the numbers are making Asterisk 1.6 (using Elastix 2) to report all
circuits are busy now. All of this numbers are cellular phones, but they
constantly reporting the same
I have cdr_adaptive_odbc compiled on my test box.
The following odbc packages are installed. I'm mainly using mysql
though so not even using this module.
We just compiled all modules in case we wanted to use them in
devel/testing.
root@telco01 /usr/local/src/asterisk-1.8.5.0 # rpm -qa |
Nigel,
I have never personally setup the Call Manager (CUCM), or whatever they
call it today, to work with Asterisk.
But I have seen what appears to be a good guide on voip-info.org.
http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integratio
n
Becca
From:
Hello
Is there a document that sums up the major changes made to the four
main releases available (1.4, 1.6, 1.8, and 10), to check if it's
worth upgrading?
www.asterisk.org/downloads
Thank you.
--
_
-- Bandwidth and
On 02/06/2012 12:25 PM, isr...@gmail.com wrote:
Your running into a bug and the only way to solve it is to report it and debug
it and hope for a fix
There is no way someone can help without it being debugged and knowing what's
causing it to lockup
The only key to unlcock it when it gets
On 12-02-06 09:15 AM, Jonas Kellens wrote:
On 02/06/2012 12:25 PM, isr...@gmail.com wrote:
Your running into a bug and the only way to solve it is to report it
and debug it and hope for a fix
There is no way someone can help without it being debugged and knowing
what's causing it to lockup
The
On 02/06/2012 03:19 PM, Paul Belanger wrote:
On 12-02-06 09:15 AM, Jonas Kellens wrote:
On 02/06/2012 12:25 PM, isr...@gmail.com wrote:
Your running into a bug and the only way to solve it is to report it
and debug it and hope for a fix
There is no way someone can help without it being
Frank
Once you have installed new dependencies you must start from scratch.
Did you do a make clean then run ./configure again and then do your make
menuselect ?
Thanks
Bryant
From: Rebecca Robinson rebecca.robin...@amgsrv.com
Sent: Monday, February
I was doing that, I think it was because the unixodbc-devel package wasn't
installed
On 6 February 2012 14:19, Bryant Zimmerman brya...@zktech.com wrote:
Frank
Once you have installed new dependencies you must start from scratch.
Did you do a make clean then run ./configure again and then do
Jonas
Try asterisk -rnx show locks /locks.txt
If that does not work than your CLI process is completely gone and you are
done. I have bumped into these kind of CLI locks and it really sucks. I
have written a program to monitor for them and kill the asterisk process if
they occur then
On Monday 06 February 2012, Frank Church wrote:
When I do a make menuselect cdr_adaptive_odbc is disabled.
What packages are required to enable it?
Even after executing apt-get install unixodbc libmyodbc odbc-postgresql
tdsodbc unixodbc-bin it is still disabled.
What am I missing?
#!/bin/bash
# checksetexternip.sh
# Author: John Cahill em...@johncahill.net
# Licence: GPL v3
# Description: script that queries checkip.dyndns.com to find the server's
external IP address. Updates asterisk's externip value and does a sip reload if
necessary.
# Last modified 06/02/2012
Just my .02 - I don't think DUNDI will be deprecated any time soon (they
tend to warn you about that at least 6 months out). I think it is in
Extended because it isn't used in a majority of installs.
From: asterisk-users-boun...@lists.digium.com
Note: You'll probably have to change /etc/asterisk/sip_general_custom.conf to
/etc/asterisk/sip.conf in the script depending on your set-up.
- Original Message -
From: John Cahill j...@dmcip.com
To: Asterisk Users Mailing asterisk-users@lists.digium.com
Sent: Monday, 6 February, 2012
On 12-02-06 09:23 AM, Jonas Kellens wrote:
On 02/06/2012 03:19 PM, Paul Belanger wrote:
On 12-02-06 09:15 AM, Jonas Kellens wrote:
On 02/06/2012 12:25 PM, isr...@gmail.com wrote:
Your running into a bug and the only way to solve it is to report it
and debug it and hope for a fix
There is no
Is there a document that sums up the major changes made to the four
main releases available (1.4, 1.6, 1.8, and 10), to check if it's
worth upgrading?
www.asterisk.org/downloads
The UPGRADE.txt and CHANGES files do just that. They have been a part
of the Asterisk source files for a long
Perhaps someone with too much time on their hands could parse the CHANGES
files and make a nice spreadsheet/PDF that puts all of the changes into a
chart form?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
I have been following the Asterisk CLI unresponsive thread and saw the
suggestion to enable BETTER_BACKTRACES
when I went in to do that it said it requires libbfd.. When I went to add
that from yast I get no packages by that name.
I am on openSUSE an ideas if this is part of another package or
You are mis-understanding the concept - the noanswer option is playing the
file as you requested, but since you aren't answering the call, no channel
is established to actually present the sound to you.
From: asterisk-users-boun...@lists.digium.com
Local/queue/?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Cabrera Obed
Sent: Monday, February 06, 2012 1:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
No, Local/queue/ don't work at all :(
2012/2/6, Danny Nicholas da...@debsinc.com:
Local/queue/?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Cabrera Obed
Sent: Monday, February 06,
Queue()?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Cabrera Obed
Sent: Monday, February 06, 2012 1:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Install binutils-devel - this includes libbfd.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Monday, February 06, 2012 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
No :(
2012/2/6, Danny Nicholas da...@debsinc.com:
Queue()?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Cabrera Obed
Sent: Monday, February 06, 2012 1:40 PM
To: Asterisk Users Mailing
In short - is this module essential for the running of Asterisk? What is
its function? Is there a help/list where I could find a description of
what it does? Thanks!
--
_
-- Bandwidth and Colocation Provided by
Why do you see binding to 0.0.0.0 to be a security risk?
Purely because a response from Asterisk can be received as a result of a
connection on *any* interface on the system/machine. If I have Asterisk
confined to, say, 2 interfaces - eth0 (10.1.1.1) and eth1 (10.2.1.1)
then a request over a
While usually thread hijacking is not something that should be done,
in this case thank you for hijacking it as the OP on his original
topic was way off topic.
Why is that - I think I posted legitimate questions/queries with regards
to the installation, configuration and running of Asterisk
Your description sounds almost entirely like the existing call
screening, so I'm pretty sure you'll be able to accomplish it. Start
with call screening, and modify that to suit your needs.
It is indeed. This is already implemented in Asterisk I take it then? If
so, brilliant news!
I'd
On Mon, Feb 6, 2012 at 8:24 PM, Josh mojo1...@privatedemail.net wrote:
In short - is this module essential for the running of Asterisk? What is its
function? Is there a help/list where I could find a description of what it
does? Thanks!
The primary goal was to upload audio for IVRs in the
Are there any ATAs that support IPv6 in the wild, given that IP4 address
are running out?
/voipfc
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
From: David Klaverstyn da...@klaverstyn.com.au
To: Asterisk Users Mailing List - Non-Commercial Discussion
(asterisk-users@lists.digium.com) asterisk-users@lists.digium.com
Sent: Monday, February 6, 2012 5:50:59 AM
Subject: [asterisk-users] Asterisk 10 and DUNDi, Extended Support?
When I
- Original Message -
From: Jonas Kellens jonas.kell...@telenet.be
To: isr...@gmail.com, Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Monday, February 6, 2012 8:15:50 AM
Subject: Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive
On Tuesday 07 Feb 2012, Josh wrote:
[snip]
Unfortunately, (IIRC) Asterisk does not reply to the same interface
packets are received from which limits the usefulness of multiple
interfaces.
What do you mean by that? If a request is received over eht1 are you
saying that Asterisk does not
On Monday 06 Feb 2012, John Cahill wrote:
logger -s checksetexternip.sh: External IP address
has changed, changing /etc/asterisk/sip_general_custom.conf grep -v
externip /etc/asterisk/sip_general_custom.conf
/etc/asterisk/sip_general_custom.conf.tmp echo externip=$EXTERNIP
To me it would be simpler to use externhost instead of externip and then use a
dynamic DNS service. It has worked flawlessly for me for many years.
Regards
David.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Hey,
I've heard recently from quite a few customers that there's cordless handsets
around which don't require a lifter.
Is anyone aware of any of these which will work with the cisco 69xx's, 79xx's
or any of the current polycom range?
-Blake
--
On 02/06/2012 03:29 PM, Josh wrote:
Your description sounds almost entirely like the existing call
screening, so I'm pretty sure you'll be able to accomplish it. Start
with call screening, and modify that to suit your needs.
It is indeed. This is already implemented in Asterisk I take it then?
On 02/06/2012 03:27 PM, Josh wrote:
Why do you see binding to 0.0.0.0 to be a security risk?
Purely because a response from Asterisk can be received as a result of a
connection on *any* interface on the system/machine. If I have Asterisk
confined to, say, 2 interfaces - eth0 (10.1.1.1) and eth1
Thanks for this explanation Dany!
Regards,
Zohair Raza
On Mon, Feb 6, 2012 at 10:11 PM, Danny Nicholas da...@debsinc.com wrote:
You are mis-understanding the concept – the noanswer option is playing the
file as you requested, but since you aren’t answering the call, no channel
is
Hey Danny,
I've this thing exactly running and working as Zohair mentioned! i.e I do
not answer() the call rather put a progress() and soon after that playing
back the sound file from playback with noanswer and then I get the file
streaming as 183-Session progress file.
I do understand that
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