Hi,
We're located in London (Ont) but have setup asterisk for
clients from Mississauga to Windsor (as well as lots of international clients
assisted remotely). We've also taken on roles from one extreme to the
other (i.e. deliver a turnkey solution including hardware, to just giving advice
Sorry
all - that was mean to go directly to the user, not the whole list
:)
From:Sent: Friday, September 02,
2005 7:54 AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: RE: [Asterisk-Users] Any one in Toronto / Canada
can help me!
Hi,
We're located in London (Ont)
Take
a look at www.generationd.com -
they have a free script for setting up ISA 2004 for SIP access (running on same
box as ISA). They have experience setting up ISA Asterisk for
customers (but charge for support since that's their
business)
From: Dean Collins [mailto:[EMAIL PROTECTED]
I have FC4 working well!
-Original Message-
From: Asterisk Supporter [mailto:[EMAIL PROTECTED]
Sent: Friday, August 26, 2005 1:16 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Fedora Core 4 x86_64
I am about to build a Dual Opteron Asterisk box as our soon to be
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Make asterisk 1.0.7 fail under FC4
In article [EMAIL PROTECTED],
Michael Stahl [EMAIL PROTECTED] wrote:
After more investigation, I decided to just recompile asterisk (on my
newly upgraded Fedora core 4 system). Make dies
I just upgraded to Fedora Core 4 and Asterisk won't run any more. When
launching asterisk, I get asterisk: error while loading shared
libraries: libssl.so.4: cannot open shared object file: No such file or
directory.
A quick search (find / -name libssl.so.4) for the file shows the file
nowhere
After more
investigation, I decided to just recompile asterisk (on my newly upgraded Fedora
core 4 system). Make dies with this error:
"No rule to make
target
'usr/lib/gcc/i386-redhat-linux/3.4.3/include/stddef.h"
It seems this
directory is gone under FC4, and replaced by
No rule to
Use a script to rotate the logs every night (by cron). If you need one,
I'll post what I have.
That's only have the solution of course...you would have to write
another script to delete logs X days old.
Anyone have one of those handy?
-Original Message-
From: Leo Burd [mailto:[EMAIL
Take alook at thevia arena web
site. Your processor may look like a 586 to the installer but may not
support all of the instructions (causing a crash). The via arena site
gives instructions on how to compile and get it installed on your
processor!(I have the C3 Nehemiah processor so I
I went with Fedora - great support and eas of use (because of Red Hat
shared tools).
So far so good!
-Original Message-
From: Scott Kamp [mailto:[EMAIL PROTECTED]
Sent: Sunday, July 03, 2005 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
The system startup script /etc/init.d/asterisk calls the script
/usr/sbin/safe_asterisk
In safe_asterisk, the program is started with -c by default (console on
TTY9).
That explains why it is starting with a console, but not why it's
running so many times! Here is what my system (FC3) shows:
You should be able to do a good job with IPTABLES which is included in
FC3. You can limit source destp IP and protocol, etc.
Type man iptables | more for more details...
OCG
-Original Message-
From: Terry H. Gilsenan [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 30, 2005 8:11 PM
I haven't heard much feedback yet - anyone here using VocTel?
The connection problem turned out to be my firewall, but I'm curious if
others experience any voice choppiness or high latency. Some posters
have related the problem to specific VOIP providers, some seem to be ISP
related (local
I have Fedora Core 3 running great on an Epia
mobo
From: Wiley Siler
[mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 12:54
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: [Asterisk-Users] Epia C3
Linux
Anyone know a good distro for an
Epia Mobo with the C3
It installed directly from the FC3 dvd, no changes...no
external drivers required
From: Wiley Siler
[mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 2:42
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: RE: [Asterisk-Users] Epia C3
Linux
Did it require any
If your phones are setup to connect to the asterisk box by
name, then a smart DNS server can just point phones to the backup box after
failure. However, since asterisk running on the backup box doesn't know
about the phones, this is only half the solution
From: Mohamed A. Gombolaty
I have an analog phone connected to an ATA
connected to asterisk. Isthere a way to access features like FORWARD and
PARK - with a call already active (on the analog line)
Thanks,
OCG
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Post your iptables output (iptables -nvL) and I'll send you the
commands you need
OCG
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, June 29, 2005 9:17 PM
To: asterisk-users@lists.digium.com
Subject: Asterisk-Users
Any users of the
VocTel VOIP service? (Canadian)
How have you found
the quality (Choppy / smooth audio)?
Any problems
registering? (I have been unable to register for
hours)
After reading about
the collapse of a big USA VOIP provider, I'm curious
Thanks,
OCG
These linksys boxes may be configured to ignore ICMP/PING
on the WAN side - so your connection may be ok! Go to the config web pages
of the router to enable PING.
OCG
From: Dionisis Koumouras
[mailto:[EMAIL PROTECTED] Sent: Monday, June 27,
2005 6:36 AMTo:
Has anyone written a
LogWatch script for Asterisk? I use logwatch for monitor all my critical
services and would like to do the same for Asterisk.
LogWatch is very
popular, so I'm guessing that someone has created one but hasn't had time to
post it somewhere...
Thanks,
OCG
I have a fresh
install of asterisk on a Fedora FC3 machine. (I'm new to unix so forgive
the next question if it's obvious).
Why does "ps ax"
show 6 copes of asterisk running?
3338
? S 0:00
asterisk -vvvg -c3370 ?
S 0:00 asterisk -vvvg -c3372
? S 0:00
asterisk -vvvg -c3374 ?
S 0:00
I have a SPA-2001 and I didn't realize I could use calling
features on an analog handset. Does that mean you can dial *77 and use a
VOIP feature? (like forward or hold)?
Mike
From: Jorge Carrasquillo
[mailto:[EMAIL PROTECTED] Sent: Thursday, June 23, 2005
12:52 PMTo: Asterisk Users
I have a new
asterisk install (1.0.7) - and in case it's relevant I'm not using autoload
option in modules.conf. Generally all is working well. However, when
I make a call from my softphone and try to leave a message, the message is
cutoff after a few seconds (whenever I pause for 1 second
Is is possible to
use any of the advanced VOIP features from an analogphone on the analog
side of the ATA (like a SPA-2001)?
I'd like to use
hold, transfer, and conference, etc.
Thanks
___
Asterisk-Users mailing list
If I understand correctly what you are trying to do, I would suggest you set a
default context in your SIP.CONF file. That context would be an entry point
into the dialplan that is secure (eg: no outside access) that can also forward
directly to a single extensions.
Mike
-Original
I have an a problem
with audio between phones (if both are SIP, or one is SIP andthen AIX to the PSTN phone). When music on hold plays it is
choppy, and the console always shows messages like:
monmp3thread XXX
bytes of audio while expecting
What is the cause /
what can I do to
I had my asterisk
server working fine with FWD as a SIP provider, so I now added a second SIP
provider (voctel). The addition to my sip.conf file is almost identical to
FWD, however, asterisk now generates lots of debug messages for some strange
reason! In particular, the line "#
For some reason,
Asterisk is reporting the wrong timezone on voice messages. I found the
tz= parameter to add to each mailbox in voicemail.conf, but I would rather set
it once for all of asterisk.
Is there a single
location to set timezone as the default for all of asterisk?
Mike
I have the voicemail
format set to wav49 in my voicemail.conf file. When retrieving voicemails,
the first message plays back ok - but then Asterisk hangs up and the log shows
the following error. Any idea what's up?
May 19 12:57:24
VERBOSE[7860]: Asterisk Ready.May 19 13:48:51
I'm thinking of
placing Asterisk on an itx motherboard in a tiny case. The ITX
motherboards top out around 400Mhz PII (in terms of power relative to a
desktop).
How much CPU would I
need for an office of 50 people? How much disk storage for voicemail +
OS? (typical / average)
The system
My new asterisk
install seems to be running fine - including playing all prompts etc without
error. However, when placing someone on hold they here choppy music (first
second or so) then quiet. I see the errors below.
What is causing
this? (Note that I am running AsteriskWin32).
Thanks,
I'm using the default mp3 files that ship with
Asterisk:
fpm-calm-river.mp3
fpm-world-mix.mp3
If they were variable bit rate, I think I would see a
warning about 'varibel' or similar...is anyone else able to get these files to
work?
-Mike-
From: Sander [mailto:[EMAIL PROTECTED]
Sent:
I bought my ATA from them. The purchase went well,
but after the item arrived I had a concern.
The item appeared to have a problem - but no one at
VOIPSUPPLY answered their phones. I left voice messages but no one called
back. I finally got tech support from another user on their web
site.
My dial plan seems
to work great - in that when I call extensions 1234 it connects to 1234.
Strangely, after the call terminates (the other side hangs up first), Asterisk
continues in the same context and then matches to extensions _. which causes an
invalid extension error!
Why does
Although getting Linux running on the XBox sounds easy -
it's not. In particular, if you get a ver 1.6 box you will be hard pressed
to get Linux installed. Be prepared to spend $ on a boot chip, a
replacement DVD drive (unless you are lucky to get one of the samsung drives),
an upgraded
I'm using EyeBeam
from xten, and whenever I call another user, the callee phone rings but my SIP
phone immediately hangs up. The other end keeps on ringing but when the
callee answers, there is no sounds.
I have found the
"Didn't get frame from channel" error occurring in each such call.
My home asterisk seems to work- I can call from
one internal phone to another. However, just leaving my system idle always
generates an error message relating to a NOTIFY. See the log below.
Any ideas?
Thanks,
Mike
--MESSAGE
FILE-
to
I just bough a
Sipura SPA-2100 to use with Asterisk. When I use the analog handset
plugged into the SPA-2100, the person on the other end can hardly hear
me.
I check the SPA-2100
setup and their is no mic/spk gain control. Is this a problem with the
SPA-2100 or with Asterisk? Any way for
search every config page (in advanced mode) and can't see
it.
Thanks,
Mike
Michael Stahl wrote:
I just bough a Sipura SPA-2100 to use with Asterisk. When I use the
analog handset plugged into the SPA-2100, the person on the other end
can hardly hear me.
I check the SPA-2100 setup
I have asterisk up and running now, and installed XLITE on 2 PC's. Both
machines (mistakenly) registered as the same user / extension.
Strangely, asterisks allows this and both phones can make calls! But,
only the first one to register can receive calls at the extensions.
1. Is this normal
I have asterisk up and running now, and installed XLITE on 2 PC's. Both
machines (mistakenly) registered as the same user / extension.
Strangely, asterisks allows this and both phones can make calls! But,
only the first one to register can receive calls at the extensions.
1. Is this normal
I have setup my
rtp.conf to only user ports in the range 1-10005. However, my firewall
log shows asterisk is still trying to reach ports 12772 (an others). Even
after restarting asterisk.
1. Does the RTP.CONF
file define RTP ports for INBOUND, OUTBOUND, or both
directions?
2. Assuming
I have asterisk working great on my LAN but cannot
open the necessary RTP ports on my ISA firewall server to get to the outside
world. (I must open each port manually due
to an ISA limitation, and opening 1 ports will takes weeks).
Ideally, I would like to reduce the number of RTP ports
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