Hi,
this is the outpu to df -h command:
root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
S.ficherosTam. Usado Disp. % Uso Montado en
/cow 14G 4,5G 8,7G 34% /
udev 999M 4,0K 999M 1% /dev
tmpfs 403M 860K 402M 1%
Take a look here:
http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device
Am 06.03.2013 13:00, schrieb termo termosel:
Hi,
df -h output:
root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
S.ficherosTam.
Try to set the tmp variable. In your case:
mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make
Am 06.03.2013 13:20, schrieb termo termosel:
Hi,
I read it but I don't find the solution. How Can I alocate more free
space in tmp?
Thanks,
Jordi
Hi,
the same error, I write your commands:
mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make
but the same error happens
/usr/bin/ld: final link failed: No space left on device
collect2: ld devolvió el estado de salida 1
make[2]: *** [asterisk] Error 1
make[1]: *** [main] Error 2
Did you execute the make command in the same environment so that make
really uses the TMPDIR directory? (no su or other shell)
Am 06.03.2013 13:37, schrieb termo termosel:
Hi,
the same error, I write your commands:
mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make
but the same error
You'd probably be better off sending this to the dev list (asterisk-dev)
Justin Killen
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Optical Phoenix
Sent: Tuesday, March 05, 2013 5:56 PM
To: asterisk-users@lists.digium.com
Subject:
Thanks, will do
On Wed, Mar 6, 2013 at 11:24 AM, Justin Killen
jkil...@allamericanasphalt.com wrote:
You’d probably be better off sending this to the dev list (asterisk-dev)*
***
** **
Justin Killen
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
Hi every body,
Please if some one could help me with this:
I'm writing an AGU Perl Script which basically makes a call using an
extension provided by other asterisk box to an E1. The asterisk version is
1.6.0.28, so it hasn't the Wellington know AGI class. The code is as follows:
I'm going to make an observation here that may upset you, and I don't mean
it to, but it's fact. If you are so unfamiliar with Linux, you will have a
bad time managing Asterisk servers. You really need to know how to use the
OS before you can learn to manage services running on it. I strongly
Solved.
2013/3/5 Luis H. Forchesatto luisforchesa...@gmail.com
Greetings.
I got two asterisk servers, one is connected to another via sip trunk. The
incoming calls are routed to the time period an if matches is transfered to
the designed extension. If don't, is redirected to a second time
Couldn't agree more, Carlos. But then again, haven't we all started this
way? ;-) The best way to understand Linux is learning the hard way. After
all, it takes a genius to understand the simplicity of Linux.
Sent from my iPhone
On 6 mrt. 2013, at 17:53, Carlos Alvarez car...@televolve.com
Might be a codec issue, try allow=all in your sip.conf
Sent from my iPhone
On 6 mrt. 2013, at 17:49, Gustavo Salvador
gustavo.salvador...@gmail.com wrote:
Hi every body,
Please if some one could help me with this:
I'm writing an AGU Perl Script which basically makes a call using an
On Wed, Mar 6, 2013 at 10:02 AM, Gertjan Baarda gertjan.baa...@gmail.comwrote:
Couldn't agree more, Carlos. But then again, haven't we all started this
way? ;-) The best way to understand Linux is learning the hard way. After
all, it takes a genius to understand the simplicity of Linux.
If
Thanks,
But SIP uses the caller box to send the call to the second box where is running
the AGI script, the second box uses DAHDI to routes the call to E1. I've tested
the codec routing a call between a E1 extension and a local one with the
originate extension command and works.
So that is
On Wed, 6 Mar 2013, Gustavo Salvador wrote:
I'm writing an AGI Perl Script...
=
#!/usr/bin/perl
use strict;
my %AGI;
:
print EXEC Dial(DAHDI/g2/$AGI{dnid},,W);
=
Is this your entire script or just a snippet? If this is all, this is
El 06/03/13 11:52, Carlos Alvarez escribió:
I'm going to make an observation here that may upset you, and I don't mean it to, but it's fact. If you are so unfamiliar with Linux, you will have a bad time managing Asterisk servers. You really need to know how to use the OS before you can learn to
Hi,
I am running asterisk 1.8.14.0, It was running fine for last few days and
suddenly crashed today
In logs I can see that abrt tried to save the core dump but it couldn't
Mar 6 12:11:09 localhost kernel: asterisk[26544]: segfault at 72656d69ac
ip 00533c19 sp 7f7db9ce3af0 error 4
Thanks Steve,
That was just a snippet, the complete script is as follow:
#!/usr/bin/perl
use strict;
$|=1;
# Setup some variables
my %AGI; my $tests = 0; my $fail = 0; my $pass = 0;
my $key; my $value;
while(STDIN) {
chomp;
last unless length($_);
if
Please don't top-post...
On 06/03/2013, at 13:24, Steve Edwards asterisk@sedwards.com
wrote:
Can you enable AGI debugging on the Asterisk console and see if that
yields any clues?
On Wed, 6 Mar 2013, Gustavo Salvador wrote:
I have run asterisk in verbose mode, and also set on the
Le 06/03/2013 17:57, Luis H. Forchesatto a écrit :
Solved.
Great, happy for you.
What would be nice is to explain how you solve it for archives. Other
people can run in the same problematic that yours and would be happy to
see your way to get out of it
2013/3/5 Luis H. Forchesatto
On Wed, Mar 6, 2013 at 3:48 PM, Administrator TOOTAI ad...@tootai.net wrote:
Le 06/03/2013 17:57, Luis H. Forchesatto a écrit :
Solved.
Great, happy for you.
What would be nice is to explain how you solve it for archives. Other people
can run in the same problematic that yours and would
El 06/03/13 17:48, Administrator TOOTAI escribió:
Le 06/03/2013 17:57, Luis H. Forchesatto a écrit :
Solved.
Great, happy for you.
What would be nice is to explain how you solve it for archives. Other
people can run in the same problematic that yours and would be happy
to see your way to
Can you provide OS details ? Its seems problem of abrt. Did u tested
asterisk without abrt
Regards,
Bharat Lalcheta
On Thu, Mar 7, 2013 at 12:05 AM, Zohair Raza
engineerzuhairr...@gmail.com wrote:
Hi,
I am running asterisk 1.8.14.0, It was running fine for last few days and
suddenly crashed
Hello, We need to setup asterisk server for 1000 extensions and in this setup
only extension to extension dialling is required (without call recording and
voicemail), like intercom calling. Please let us know what can be the best
economic solution/setup for this. Thanks,Kamlesh
I am still facing this issue. Is AsteriskNOW and the CentOS repositories
depreciated?
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Steven Howes
Sent: Wednesday, February 06, 2013 9:28 AM
To: Asterisk
On Thu, 7 Mar 2013, Kamlesh Kumar wrote:
We need to setup asterisk server for 1000 extensions and in this
setup only extension to extension dialling is required (without call
recording and voicemail), like intercom calling. Please let us know what
can be the best economic solution/setup for
Technology is SIP and asterisk is not handling the media, what is cheapest
solution to be used for SIP client. Thanks,Kamlesh
Date: Wed, 6 Mar 2013 20:43:52 -0800
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk with 1000 extensions
On
On Thu, 7 Mar 2013, Kamlesh Kumar wrote:
Technology is SIP and asterisk is not handling the media, what is
cheapest solution to be used for SIP client.
Client? How about a free SIP softphone?
Server? How many calls per second? How many simultaneous calls? Any
half-way recent box should do.
softphone is not going to be used in this setup. Hardphone is required. Around
60-70 simultaneous calls would be required. Thanks,Kamlesh
Date: Wed, 6 Mar 2013 21:15:51 -0800
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk with 1000
Please don't top-post.
On Thu, 7 Mar 2013, Kamlesh Kumar wrote:
softphone is not going to be used in this setup. Hardphone is required.
Around 60-70 simultaneous calls would be required.
OK. So figure on about 6 UDP packets, about 3.5 KB per call. Not a big
deal.
I'd look for a reliable
Server side installation with recent hardware is fine, we can build two
parallel system for redundancy. We are more concern with the cost of SIP client
(hardphone). What are the various ways to make this setup functional with low
cost for SIP clients. Thanks,Kamlesh
On Thu, 7 Mar 2013,
Please don't top-post.
On Thu, 7 Mar 2013, Bharat Lalcheta wrote:
You can use ATA box with pstn phone to reduce cost.
Are you wiring a building where multiple-line SIP gateways make sense?
How about a description of what you are trying to do?
Personally, I like Polycom SIP phones but I
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