2009/8/12 Jonathan Thurman jthurma...@gmail.com
On Wed, Aug 12, 2009 at 12:39 PM, Olivier oza-4...@myamail.com wrote:
2009/8/12 Jonathan Thurman jthurma...@gmail.com
I am also using them quite extensively, but with English menus. I know
that the Locale files from Cisco do not come with
Hi All,
I had a problem yesterday, that our asterisk server showed 2 channels were
in use continuously, but nobody were using any of them at that time. I had
to kill them using softhangup and I checked all the logs but could not
find why exactly this problem occurred, as the system was running
On 13 Aug 2009, at 07:51, das sandesh wrote:
Hi All,
I had a problem yesterday, that our asterisk server showed 2
channels were in use continuously, but nobody were using any of them
at that time. I had to kill them using softhangup and I checked
all the logs but could not find why
Summary: Is it possible to append to a user-defined text file from the
dialplan without using AGI ?
Background: We are using our Asterisk server to place outbound calls
alone. We've 8 E1s that enables us to place about 240 calls
concurrently. The server is a bit old, and we are not able to reach
Why don't you setup a second server just for mysql and fastagi.
Fastagi can also be installed on your asterisk server with just a small
overhead, as fastagi will keep running in the background, you will not
have the overhead of creating new process at each call.
Olivier
Unni a écrit :
would somebody be able to recommend a good package that works with Asterisk
please ... not commercial as will be mainly used for my home office.
Best Regards,
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Hellos,
I am having issues with my meetme conferencing. When I dial the conferencing
number, It hangs after a few seconds.I have read somewhere that I need to
enable ztdummy, which I have done but still no changes.
Here is my log
~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.08.12 18:44:43
Thanks for the pointers guys!
S.
- Original Message -
From: Dave Fullerton dfullertaster...@shorelinecontainer.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, August 12, 2009 1:20:11 PM GMT -08:00 US/Canada Pacific
Hi all,
I was looking to build a SIP-to-PRI gateway using Asterisk (as in my other
post), but there is also an alternative of using a Cisco router with something
like an NM-HDV module with a T1 VIC module and DSP channel banks.
The question is, would it be more reliable to offload all
Shashi Dookhee wrote:
The question is, would it be more reliable to offload all dahdi/zaptel/libpri
type stuff to a dedicated gateway device (Asterisk or Cisco)
Yes - for Cisco. No - for Asterisk.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678)
On 13 Aug 2009, at 11:37, Alex Balashov wrote:
Shashi Dookhee wrote:
The question is, would it be more reliable to offload all dahdi/
zaptel/libpri
type stuff to a dedicated gateway device (Asterisk or Cisco)
Yes - for Cisco. No - for Asterisk.
Asterisk can do it but its pretty much
I've written another round up article on soft phones that you can use
with Asterisk.
I've tried to omit any that required you to sign up before downloading,
or had time limits to usage etc.
There are a couple in there that actually have their own servers but may
be interesting nonetheless.
I
On Thu, Aug 13, 2009 at 6:25 AM, Shashi Dookhee sdook...@fortify.comwrote:
Hi all,
I was looking to build a SIP-to-PRI gateway using Asterisk (as in my other
post), but there is also an alternative of using a Cisco router with
something like an NM-HDV module with a T1 VIC module and DSP
On 13/08/09 8:33 PM, Unni wrote:
Summary: Is it possible to append to a user-defined text file from the
dialplan without using AGI ?
You could use inotify to watch a directory owned by your webserver, then
when you add a new rate to it cause it to run a script that updates the
dialplan and
Hi
Someone may have an issue to my problem :
- I install mysql server and create a database named asteriskdb to store
some data of asterisk. OK
- I create a table cdr in order to replace de Master.csv . OK
- I set up Areski CDR (asterisk-stats) for my environnement thank to
The appropriate venue for this question is some forum, mailing list or
other communication channel provided by the creators of Areski.
This is not an Asterisk issue.
harry R wrote:
Hi
Someone may have an issue to my problem :
- I install mysql server and create a database named asteriskdb
Thanks Steve.
The channels which got stuck up are the SIP channels.
If I use rtptimeout then if there is some silence at both the ends the
call may get disconnected after that period of time also if I use
Absolutetimeout then there are chances that some calls may be upto 30min
to 40min, if our
Hello Mailinglist,
i was reading a paper regarding a Asterisk clustering solution and
they where pretty excited about a feature in polycom phones:
You can add a registration to a primary asterisk server
You can add a registration to a secondary asterisk server
The polycom phones will talk to
Have you configured your /etc/asterisk/cdr_mysql.conf file?
Ish
harry R wrote:
Hi
Someone may have an issue to my problem :
- I install mysql server and create a database named asteriskdb to
store some data of asterisk. OK
- I create a table cdr in order to replace de Master.csv . OK
- I
I linked to Areski's CDR Stats (which I've used a few times):
http://www.areski.net/asterisk-stat-v2/about.php
Asterisk-Stat is a visualisation layer for Asterisk CDR statistics which
are pulled from a database. It provides graphs as well as allowing you
to get more information on
2009/8/13 Ishfaq Malik i...@pack-net.co.uk
Have you configured your /etc/asterisk/cdr_mysql.conf file?
Yeah
I configure it. Now everytime I do a call, a CDR line is added in my table
cdr.
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I have googled everything i can on SDP and adding the codec. i can't find
anything. do you think something went wrong with the install because this seems
like a strange problem to have.
From: sjo...@ftdata.com
To: asterisk-users@lists.digium.com
Date: Wed, 12 Aug 2009 16:19:33 -0400
Hello everybody
I have an asterisk with an integration of alcatel pbx, by sip trunk, all
calls are fine, but tha calls calls that originate from a analog line,
the recipient is not listening, and that if they hear the call originates,
the lines are E1 in alcatel pbx.
When a asteris user call to
Hello,
In your sip.conf
You need
host=sip.xxx.com
or IP
don't work with dynamic
Regards
On Wed, Aug 12, 2009 at 8:27 AM, harry R rhm.noa...@gmail.com wrote:
Dear all,
I want to setup the incoming calls, that don't use authentication in
sip.conf file.
My configurations as
Check out the Failover Identity (Ersatz Identität) in the identity
settings. Works a little bit different, but you can achieve the same effect
with this.
CS
-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im
I was reading in the documentation about the SNOM phones (mainly 300)
but I did not find anything in the users-pdf's or on there
knowledgebase/website which would tell me if this is possible, there
is something for failover configuration but it is not explained at all.
It's highly appreciated
Hi Raimund,
snom uses basically the same concept. As explained under:
http://wiki.snom.com/Settings/user_failover_identity.
You select the line id that should be used when a registration fails.
Regards,
Usman
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I am seeing some curious behaviour with a 1.2.32 system, which I do not
understand and so can't work out how to fix it.
I have a PRI routed to context default. Here is the complete default
context:
[default]
exten = _9X.,1,Dial(IAX2/m1peer/${EXTEN:1})
exten = _20XX,1,Dial(IAX2/sipeer/${EXTEN})
Thank-you for your email. I will be out of the office from Thursday, August 13
until Monday, August 17. I will have limited access to email during this time
and will respond as soon as possible.
If you need immediate assistance, please call our support line at 770-674-3900
x 1.
Thank you.
I posted a message to this list about 50 minutes ago. I received a
posting acknowledgement pretty quickly, and it showed up in the mailman
list archives, but I still have not received a copy.
Looking at some of the other recent messages I have received, they have
also suffered a delay of 20
Tony Mountifield wrote:
I posted a message to this list about 50 minutes ago. I received a
posting acknowledgement pretty quickly, and it showed up in the mailman
list archives, but I still have not received a copy.
Looking at some of the other recent messages I have received, they have
also
My messages go through rather quickly (minutes).
Unless the lists.digium.com server is running on an Atari, it's probably NOT an
overload issue...
-Dave
snip
Are there any plans to beef up the mailing list server so that messages
can get through with less of a delay?
/snip
so i added the following to sip_custom.conf
allow=gsm
allow=h261
allow=h263
allow=h263p
videosupport=yes
and this to sip_nat.conf
localnet=192.168.1.0/255.255.255.0
externhost=pbx.DOMAIN.com
externrefresh=10
fromdomain=DOMAIN.com
nat=yes
qualify=yes
canreinvite=no
now it works. so it was my
In article h61ftl$nu...@softins.clara.co.uk,
Tony Mountifield t...@softins.clara.co.uk wrote:
I am seeing some curious behaviour with a 1.2.32 system, which I do not
understand and so can't work out how to fix it.
Well, after some looking at pbx.c and some pri debugging, I discovered
what the
Hello,
So much keeps changing with the dialplan and Realtime lookups. Just
downloaded the latest stable 1.6.1.2. The app_realtime, which was
perfectly brilliant and did exactly what I needed, is gone; replaced
with func_realtime. The REALTIME function is unacceptable:
; Get the conference
In article
f9e3e06bfdeee842b83287cf325faa8bd469f20...@strongbad.central.videon-central.com,
David Gibbons d...@videon-central.com wrote:
My messages go through rather quickly (minutes).
Unless the lists.digium.com server is running on an Atari, it's probably NOT
an overload issue...
We recently completed a full migration of our office phone system over
to Asterisk and are using flowroute Broadvoice as SIP providers for
our incoming/outgoing calls. Everything is working great.
Now that the phones are done, I'm left with the fax machine. Although
we have been using jFax
Hi everybody
I have a logic question that is confusing me.
ifTime(00:00-12:00|*|*|*) {
Playback(welcome-morning);
} else {
ifTime(12:00-18:00|*|*|*) {
Playback(welcome-afternoon);
My Assumption would be that 00:00-12:00 actually covers midnight to
12:00:59. I would verify this by hitting this dialplan at 30 seconds after
noon local time. Since Im still on 1.4, this one is academic to me.
_
From: asterisk-users-boun...@lists.digium.com
On Thursday 13 August 2009 14:21:42 Boehm, Matthew wrote:
How is that possible that a 10 month old patch to the trunk is not in
the most recent stable?
The branch that you're using was created before that commit. If you
download the 1.6.2.0-beta4, you'll find that the new functions are in that
Well, our team wants to go production tomorrow and don't want to do that
on non-stable code. I grabbed the patch from that bug and applied it to
1.6.1.2 func_realtime.c and seems to be working fine now with HASH
function too.
Thanks,
-Matthew
-Original Message-
From:
On 14/08/09 2:19 AM, harry R wrote:
I linked to Areski's CDR Stats (which I've used a few times):
http://www.areski.net/asterisk-stat-v2/about.php
Asterisk-Stat is a visualisation layer for Asterisk CDR statistics which
are pulled from a database. It provides graphs as well as
On 14/08/09 2:21 AM, harry R wrote:
2009/8/13 Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk
Have you configured your /etc/asterisk/cdr_mysql.conf file?
Yeah
I configure it. Now everytime I do a call, a CDR line is added in my
table cdr.
If you're still unable to see
Dáibhéad Antoine O'Reilligh wrote:
I have a logic question that is confusing me.
ifTime(00:00-12:00|*|*|*) {
Playback(welcome-morning);
} else {
ifTime(12:00-18:00|*|*|*) {
On Wed, 12 Aug 2009, Shashi Dookhee wrote:
I'd like to setup a really lean Asterisk installation that essentially
has a full ISDN PRI (ATT, T1, 23 B-chans, 1 D-chan, BZ8S, 5ESS,
National dialplan) on a Digium TE207P adapter that all it does is
convert the ISDN channels to SIP/IAX channels.
Hi,
When I am debuggin any peer, I get swamped with those messages:
chan_sip.c:8866 check_auth: Correct auth, but based on stale nonce received
from xyz
This seem to make up around 2Mbits/s of data (estimated after a quick
tcpdump).
Any ideas? This is while the server is idle (no
Sorry, that is running 1.4.26.1.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, August 13, 2009 23:22
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Stale auth
I try to off-load specific tasks like PRI-to-SIP to dedicated hardware for
the task. It is also easier to have centralized call processing and easy to
configure/manage devices in our remote locations. I have colleagues that
use Digium PRI cards just fine. Just depends on your budget and
On Thursday 13 August 2009 19:18:44 Edwin Lam wrote:
Dáibhéad Antoine O'Reilligh wrote:
I have a logic question that is confusing me.
ifTime(00:00-12:00|*|*|*) {
Playback(welcome-morning);
} else {
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