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 Alejan
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
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
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
Sent
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
Local Channel Crash Problem 1.8.3-rc2
On Tue, Feb 15, 2011 at 10:15 AM, Nic Colledge 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. I
don&
Local Channel Crash Problem 1.8.3-rc2
On Tue, Feb 15, 2011 at 10:15 AM, Nic Colledge 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. I
don&
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 Ro
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: asterisk-users-boun...@lists.digium.c
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 kee
At the asterisk CLI type "module show like timing"
Whichever has a use-count >1 is the one you are using.
Nic.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: 13 May 2011 16:03
To: tbs...@gmail.com; asterisk-users
Su
nd this?
Thanks in advance.
Regards,
Dr. Nic Colledge
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rcial 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
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 varia
. I'm currently installing 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 advan
ating 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 Co
:asterisk-users-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
From: 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 this
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/aster
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). Fo
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 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/100&SIP/outboundA/101&SIP/outboundB/200&SIP/outboundB/201)
this is r
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: asterisk-u
soon as 1.8 is stable enough.
Thanks
Bryant
____
From: "Nic Colledge"
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 differen
trunk) version from ages ago.
When I use the PGSQL native driver instead of ODBC with the same postgresql
table it works fine.
Is this a bug?
P.s. great work on 1.8.0 by the way, thanks to all the developers, testers and
everyone involved.
Thanks,
Nic Colledge
n...@njcolledge.net<mailt
15:22:53] ERROR[695]: netsock2.c:94 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
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
[mailto:aster
users-boun...@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 wrote:
> So
Paul,
I applied your patch to 1.8.0 and I'm happy to report it has fixed the problem
I was experiencing.
Thanks again.
Nic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: 23 October 2010 22
should be set to null.
Name/UsernameHost 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...@lis
, Nic Colledge 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/
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
Subject
Paul,
Thanks, just done a quick test and that looks to have fixed it.
Nic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge
Sent: 28 October 2010 09:54
To: Asterisk Users Mailing List - Non
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