Use lowercase for ftp:// . That might be the issue but it should be
easy to test. Do your FTP server logs shpw anything?
I will double check but I believe that lower case ftp is being used. I do
have upper case PlcmSpIp:PlcmSpip as the password. I will see if lower case
usernames and
Hello,
everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I
get the following :
[Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username
mismatch, have 32990900, digest has 3291119600
[Sep 12 18:59:29] NOTICE[2066]: chan_sip.c:20082 handle_request_invite:
I work in a small office and have fallen into the role of network support based
on knowing enough about networking to be dangerous.
Our office is moving from DSL to a T1. Were using Asterisk as our PBX and I'm
looking for hints or resources that might help me make the transition as error
free
On Sun, Sep 12, 2010 at 10:43 AM, Richard Stuppi rich...@stuppi.com wrote:
I work in a small office and have fallen into the role of network support
based on knowing enough about networking to be dangerous.
Our office is moving from DSL to a T1. Were using Asterisk as our PBX and
I'm
On 09/12/2010 02:34 PM, Kyle Kienapfel wrote:
Really it depends on what the capabilies of dsl were assuming you are
just using both dsl and t1 as internet connections.
a dsl that has close to 1mb/sec out and 10mb/sec or so in, is going to
be pretty comparable to a t1 actually so not really
On Sun, Sep 12, 2010 at 10:05 AM, Jonas Kellens jonas.kell...@telenet.bewrote:
Hello,
everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I get
the following :
[Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username
mismatch, have 32990900, digest has
Hello all,
I have recently finished compiling and making asterisk on a fresh
slackware 13 machine. I went through and made sure to install ncurses,
zlib, openssl, libnewt, libpri, and dahdi prior to compiling asterisk.
I configured with this:
$ ./configure --prefix=/usr --enable-threads=posix
Hi
Does anyone have experience with Synway cards like SHD-240D-CT/PCI with
asterisk and SynAst driver ?
Are they any good ?
Do they really run on Asterisk ?
Thanks.
Anita Hall,
Simmortel Voice
www.simmortel.com
--
_
--
On Sun, Sep 12, 2010 at 3:41 PM, Tom Lohmuller tln6...@saintjoe.edu wrote:
Any ideas? I wasn't able to find any solutions to a first time boot on
google, mostly just segfaults that people were getting when trying
asterisks functionality.
follow the instructions in doc/backtrace.txt and open a
In terms of telephony, a T-1 can make a huge difference over DSL. DSL gives you
a lot of raw bandwidth, true, but for voice that really doesn't matter all that
much. Voice calls only take a relatively small amount of bandwidth anyway; you
can fit dozens of concurrent calls into a DSL or T-1.
On Sun, Sep 12, 2010 at 8:57 AM, colin mcdermott
colinjamesmcderm...@gmail.com wrote:
Use lowercase for ftp:// . That might be the issue but it should be
easy to test. Do your FTP server logs shpw anything?
I will double check but I believe that lower case ftp is being used. I do
have
Hi Luki an all others who answered,
Try kill -6 (i.e. SIGABRT). That usually triggers a core dump for me.
yes, that works for testing and creates a coredump.
Thank you very much for your answer!
PS: Running Asterisk under GDB unfortunately is not an option, because
it is a production system
On 12 September 2010 23:56, Thorolf Godawa nos...@godawa.de wrote:
Hi Luki an all others who answered,
Try kill -6 (i.e. SIGABRT). That usually triggers a core dump for me.
yes, that works for testing and creates a coredump.
Thank you very much for your answer!
PS: Running Asterisk under
On 11/09/10 12:44 PM, Carlos Chavez wrote:
The past few days I started having a problem with a small call center
setup. All agents use Eyebeam 1.5 to receive calls from a queue. Eyebeam is
configured to auto answer the call. The problem is that the agents claim that
they get a call
On Sun, 12 Sep 2010, Kevin Keane wrote:
What really matters is the latency, and T-1 is a huge improvement over
DSL in that area. The easiest way to measure latency is the ping time to
a server that is “close to you” Internet-wise. A DSL has latencies of
between 40ms (if it’s extremely good
On 11/09/10 2:07 AM, isca...@free.fr wrote:
Hi,
I have a problem with my IAX softphones. After a call, when the softphone
hangup, it remains unavailable for the other softphones. It can call anybody,
but can not be reached... For example, if A call B, B answer, then A or B
hangup, and C
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Sunday, September 12, 2010 5:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Moving from DSL
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