On Sun, Nov 06, 2011 at 03:50:21PM +, Gordon Henderson wrote:
On Tue, 1 Nov 2011, Nic Colledge wrote:
Have you thought about using LXC rather than OpenVZ.
+1
There are a few references to allowing guest access to timing
hardware online.
Simples. Load up the dahdi modules
Have you thought about using LXC rather than OpenVZ.
There are a few references to allowing guest access to timing hardware online.
I've only been playing with it recently and haven't used it in production yet
but plan to soon.
As for general thoughts about virtualising asterisk, I tried it in
I was wondering if these could be spoofed recently when reading the docs.
Have you tried peerip rather than recvip?
Does that give the same result?
Nic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
New Text at Bottom:
---
hi:
my current system is 1.6.2. I have dahdi hardware card. I must
disable res_timing_timerfd module or sometimes phone calls would
become silent suddenly.
I don't know the situation in 1.8. I heard that timing is still a
problem in 1.8. should I
Are you using IAX? There are some problems causing crashes for us related to
laggyness on IAX channels with 1.8 versions.
There are a bunch of problems with IAX related to
https://issues.asterisk.org/view.php?id=17521
Nic.
-Original Message-
From:
Hi,
This may be related to an issue I added to the bug tracker. Problems around
using Local Channels across realtime / non-realtime contexts in 1.8.
Nic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Naomi
Hi,
I have been having a problem with asterisk crashing when using local channels
and realtime on asterisk 1.8.3-rc2.
The example given here is I think the easiest way to reproduce this problem.
In extensions.conf I have:
[internal]
switch = Realtime/extensions/p
exten = 301,1,Answer()
exten =
and Local Channel Crash Problem 1.8.3-rc2
On Tue, Feb 15, 2011 at 10:15 AM, Nic Colledge n...@njcolledge.net wrote:
I have been having a problem with asterisk crashing when using local
channels and realtime on asterisk 1.8.3-rc2.
Nic,
I can reproduce this using the latest SVN for the 1.8 branch
and Local Channel Crash Problem 1.8.3-rc2
On Tue, Feb 15, 2011 at 10:15 AM, Nic Colledge n...@njcolledge.net wrote:
I have been having a problem with asterisk crashing when using local
channels and realtime on asterisk 1.8.3-rc2.
Nic,
I can reproduce this using the latest SVN for the 1.8 branch
Try using ${UNIQUEID} to get the unique id of the current call. That or
something like CDR(uniqueid). Forget which off the top of my head.
Nic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Paul,
Thanks, I'll try this patch later tonight.
Nic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: 28 October 2010 03:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Mask Port Status
111 (null) (D) 255.255.255.255 0 Unmonitored
Thanks,
Nic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge
Sent: 24
, Nic Colledge n...@njcolledge.net wrote:
Further to my last, I think I found another small related issue with IAX
which is generating the following error:
Do you mind collecting a debug log [1]? Having some issues reproducing this.
[1]
http://svn.asterisk.org/svn/asterisk/trunk/doc
ast_sockaddr_stringify_fmt:
getnameinfo(): ai_family not supported
[Oct 23 15:22:53] ERROR[695]: chan_iax2.c:2305 peercnt_modify: Bad address cast
to IPv4
Is this a configuration issue or something in asterisk?
Thanks in advnace.
Nic Colledge
Sorry forgot to add this into my initial email.
The same happens with phones configured in iax.conf and the Realtime database
table.
[Oct 23 16:49:52] ERROR[1220]: chan_iax2.c:8770 update_registry: Bad address
cast to IPv4 etc.
Nic.
From: asterisk-users-boun...@lists.digium.com
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: 23 October 2010 17:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.8 IAX Registration
On Sat, Oct 23, 2010 at 11:53 AM, Nic Colledge n...@njcolledge.net wrote
the developers, testers and
everyone involved.
Thanks,
Nic Colledge
n...@njcolledge.netmailto:n...@njcolledge.net
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Hi,
I use CEL or Call Event Logging in 1.8 to get a more concise picture of what
happened in a call. We use it for a bunch of stuff including billing attended
and unattended transfers differently.
If you are thinking of upgrading, it's worth a try.
Nic.
-Original Message-
From:
From: Nic Colledge n...@njcolledge.net
Hi,
I use CEL or Call Event Logging in 1.8 to get a more concise picture of what
happened in a call. We use it for a bunch of stuff including billing attended
and unattended transfers differently.
If you are thinking of upgrading
Hi,
I am using CEL to more accurate billing information with some success. However
there is an ambiguity in the CEL data when multiple destinations are specified
in the DIAL command.
For example, if I have
Dial(SIP/outboundA/100SIP/outboundA/101SIP/outboundB/200SIP/outboundB/201)
this is
Hi,
This may be no use to you if you are using 1.4 but Call Event Logging (or
CEL) that is currently in trunk should provide an easier way to do this.
All events associated with a call e.g. Answer, Hangup, Bridge start, Transfer
etc. are logged to the usual back-ends. We use postgresql via ODBC.
Hi,
I think so, maybe someone can help clarify this for me also. I have:
rtcachefriends=yes
rtautoclear=yes
in sip.conf and was under the impression that this caches the settings from the
database until a user unregisters. When they unregister the data is removed
from the cache (rtautoclear).
Hi,
The last few times I have installed trunk versions of asterisk on Ubuntu I have
seen this error after doing a make config for asterisk.
install: cannot stat `contrib/init.d/etc_default_asterisk': No such file or
directory
The init.d links then fail to work properly (e.g. /etc/init.d/asterisk
this
has been a success when the database is not updated with a new regseconds time.
Any idea as to what I've done wrong / what's going on?
Thanks in advance.
Nic.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge
Sent: 11
-boun...@lists.digium.com] On Behalf Of Nic Colledge
Sent: Monday, December 14, 2009 9:33 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk Unregisteres IAX Friend Randomly
Bump! And some more information (see below for initial problem):
This problem is intermittent, but you don't
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge
Sent: 14 December 2009 15:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Unregisteres IAX Friend Randomly
I have tried
1.6.0 to test that as well.
This only seems to happen with real-time asterisk. (I'm using Postgres for the
backend database and the pgsql driver in extconfig.conf)
Any ideas what's going on here? Is this a known issue?
Thanks in advance.
Regards,
Dr. Nic Colledge
Hi
I have been using the CHANNEL variable as a way of checking if a user is
allowed to make outgoing calls, and what their source caller ID should be
(these values are in a database).
This works all of the time with SIP and most of the time with IAX, however
sometimes with IAX the channel
.
Regards,
Dr. Nic Colledge
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Discussion
Subject: Re: [asterisk-users] GotoIfTime problem - possible bug
On Monday 23 November 2009 12:11:02 pm Nic Colledge wrote:
I'm currently doing some testing against asterisk 1.6.1.7-rc2 (keep meaning
to upgrade) and am having a problem with the GotoIfTime dial plan function.
The asterisk
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