On Wed, 2015-07-08 at 15:09 -0400, Ryan, Travis wrote:
> Asterisk13 can do native tls with each phone? Nice.
Some soft phone support TLS,
but does anybody knows a soft phone that support pkcs11?
(keys & certs stored on a smart-card)
Hans
--
On Tue, 2014-10-07 at 08:37 -0500, Don Kelly wrote:
JG confirmed that it is possible, but it has not been defined.
Without knowing what kind of instruments you are using, a possible it
would be for a party to dial a 4-digit extension number to talk to someone
internally, completing a call
On Tue, 2014-09-02 at 13:18 -0500, Khalid Touati wrote:
so it seems Asterisk Versions does not support video I guess
Used it with jitsi and linphone softphones, works just OK.
Just for testing i did a video-call on the loop-back, great test tool
for showing the influence of (limited-)
On Fri, 2014-06-27 at 22:24 +0530, Anurag Rana wrote:
iptables -I INPUT 1 -p tcp --dport 5060 -m string
--string VaxSIPUserAgent --algo bm -j DROP
You make a fundamental mistake here.
On Sat, 2014-03-08 at 20:27 +, ad...@3a.hu wrote:
My approach (in theory only, so please correct me if I'm wrong) would be
to run asterisk on multiple boxes (one each). A dedicated monitoring
box (nagios? custom scripts?) would perform frequent checks against the
boxes (one of my
On Mon, 2014-02-10 at 10:39 -0500, Tech Support wrote:
Rather than speculate, take a look at the output of top. If you're
running out of memory, shut down useless processes. You'd be surprised what
processes get started by default that you don't need. You should also check
the Asterisk logs
On Fri, 2013-12-13 at 06:20 -0600, Don Kelly wrote:
On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote:
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want
to load balance incoming calls over IAX2 trunks. If any trunk goes
down the calls traffic will be shared with
-Original Message-
From: Gergo Csibra csi...@gmail.com
Reply-to: Gergo Csibra csi...@gmail.com, Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re:
-Original Message-
From: Rafael dos Santos Saraiva rafaels...@gmail.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users]
From: virus.c...@mail.ru virus.c...@mail.ru
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] НА: asterisk-users Digest, Vol 105, Issue 40
Date: Tue, 07 May 2013 07:53:53 +0600
help
-Original Message-
From: jg webaccou...@jgoettgens.de
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] looking for a
Might have a look at tine:
http://www.tine20.org/wiki/index.php/Admins/Asterisk_integration
hw
-Original Message-
From: Steve Totaro stot...@totarotechnologies.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing
Could it be distro-related?
I have various versions of asterisk (from 1.4 upto 11.3) running
paravirtualized or HW-virtualized with XEN.
Normally i use the pre-build packages from suse, only when i want to try
a release-candidates i need them myself.
hw
-Original Message-
From: Sandeep
Hi all,
I had to re-install a new machine and noticed that by default, ip was
only listening on 0.0.0.0, thus ipv4 only. Easily changed.
However, when looking at iax.conf, I found here the same, but it looks
like iax is still ipv4 only?
If i change bindaddr=192.168.0.1 towards bindaddr=::, and
Hi all,
Finally i got hold of some bt-dongles that seems p[retty stable, the
asus-bt211.
After installing them, i rebuild 11.3-rc1 added mobile.conf (bt-addres
and blackberry address) and mobile show devices is showing me that the
BT-link is up, and remains stable up.
Seems good, but it looks
-Original Message-
From: Jaap Winius jwin...@umrk.nl
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP account registration fails after
upgrade to 1.8
Date: Fri, 22 Mar
/03/13 13:18, Hans Witvliet escribió:
Hi,
I've been looking at the list at:
http://www.voip-info.org/wiki/view/chan_mobile
But when googling of any of the known working devices, there ain't any
for sale anymore, probably replaced by more recent types.
So, anyone around here who bought
-Original Message-
From: bilal ghayyad bilmar...@yahoo.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] digium card and virualbox
Date: Sun, 10 Mar 2013 20:18:52 -0700
Hi,
I've been looking at the list at:
http://www.voip-info.org/wiki/view/chan_mobile
But when googling of any of the known working devices, there ain't any
for sale anymore, probably replaced by more recent types.
So, anyone around here who bought recently an BT-dongle that is working
with
-Original Message-
From: Carlos Alvarez car...@televolve.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk
-Original Message-
From: termo termosel fermit...@hotmail.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Error to install Asterisk
Date:
Are these 4G comaptible
-Original Message-
From: Frank fr...@efirehouse.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re:
given on eBay, I don't think they are. The listed
freqs. are worldwide GSM since the mid 90's, but not 4G
John Novack
Hans Witvliet wrote:
Are these 4G comaptible
-Original Message-
From: Frank fr...@efirehouse.com
Reply-to: Asterisk Users Mailing List - Non-Commercial
-Original Message-
From: A J Stiles asterisk_l...@earthshod.co.uk
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
-Original Message-
From: Carlos Alvarez car...@televolve.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk
-Original Message-
From: Olivier oza_4...@yahoo.fr
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] OT - Chan-mobile
On Tue, 2013-01-08 at 08:21 -0600, Danny Nicholas wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry
Geis
Sent: Monday, January 07, 2013 6:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Tue, 2012-12-18 at 10:01 -0600, Jonathan Rose wrote:
Harish Mandowara wrote:
I have Asterisk server 1.8.19 with jabber enabled.
On the other side i have openfire server with asterisk-im enabled.
I have a doubt, whether my sip client connected with asterisk can
send message to
Hi all,
I'm caught up in a struggle between people how can not make up their
mind... Half way implementing a asterisk farm and they come up with
another feature they've seen in kamaillo.
What he showed me was this: three registered sip users,
a) one changes his presence status on his softphone,
On Tue, 2012-12-11 at 23:02 +0800, Barco You wrote:
Dear List,
Where can I find a guide for setup an Asterisk server which can
eastanblish a simple video call from two sip clients?
Thank you!
Regards,
Barco
Hi Barco,
I don't think there is a specific guide for this.
From the
On Thu, 2012-11-15 at 12:13 +0100, Frederic Van Espen wrote:
On Thu, 2012-11-15 at 08:53 +0100, Hans Witvliet wrote:
In stead of 12345678 i would like to use b.c.o.gr...@minoss.nl
But afaicr the dots will cause problems
If your extension does not start with an underscore
Hi all,
Is there a simple way of disabling regular expressions in the dialplan?
Reason for asking, is that people hate to remember numbers.
So i want to use there full smtp address as as their extension.
In stead of 12345678 i would like to use b.c.o.gr...@minoss.nl
But afaicr the dots will
On Thu, 2012-11-08 at 10:07 +0100, martin f krafft wrote:
also sprach Jeff LaCoursiere j...@sunfone.com [2012.11.07.2049 +0100]:
Just to chime in, if you REALLY want multi-tenant, it is super
easy and surprisingly efficient to use kernel level virtualization
to run multiple instances of
On Thu, 2012-10-18 at 15:45 +, Mahendra Dobariya wrote:
hi,
I want to use asterisk as IVR system ,
but to make and receive GSM call, i want to use 3g usb modem.(voice
enabled)
http://www.huaweidevice.co.in/Products/MobileBroadband/E303c.php
and i want to install this system on two
On Thu, 2012-10-18 at 17:18 +0100, Steven Howes wrote:
On 18 Oct 2012, at 16:50, Mitul Limbani wrote:
U would have to write a dahdi module for this 3G modem to help
asterisk understand it as standard gsm channel.
Look up chan_datacard (i think that's what it's called from memory).
Hi,
With regards to:
On Mon, 2012-10-15 at 09:09 -0500, Joshua Colp wrote:
asterisk asterisk wrote:
Dear all,
Hola,
I wish to ask a question of the new Motif Channel in asterisk 11.
I successfully compile the binary and run without error. However, when
dialing out, no external
On Wed, 2012-10-10 at 18:09 -0300, Joshua Colp wrote:
[snip]
Yes, there is no capability for video transcoding in any version of
Asterisk.
Thanks for pointing out!
So in case my managers starts nagging about it, they have two options:
A) use hard/soft-clients with comparable codecs,
B)
Hi,
Perhaps can someone tell me if i had the wrong expectancies
If one sip-clinet only supports GSM-codec, and another only supports
g711-U, they still can call each other and asterisk does the transcoding
Correct?
If i try to do the same with an AV-call, (one only h264, the other only
Hi,
Are there any thoughts about how cpu-expensive motif is?
Does it only translate SIP -- jingle (during call-setup)
if so, impact will probably neglectible.
or does asterisk remains constantly in between the data-stream?
In that case, it might be something to pay serious attention to, when
On Tue, 2012-10-02 at 17:11 -0700, Ira wrote:
At 02:19 PM 10/1/2012, you wrote:
So respond here and let me know what you think. I got a couple of replies on
the -dev list and they said that this would be good to put out on the -users
list too.
Mark Michelson
In true Republican fashion,
On Fri, 2012-09-28 at 01:33 -0700, Vieri wrote:
Hi,
According to http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip:
[snip]
So it seems that the contrib directory and the asterisk.org wiki are
inconsistent and incomplete.
Of course I understand that these are 'contributed' files
Hi all,
For one of my inverstigations it looks like i'm back to square one
I'm trying to accept an incoming xmpp call and forward it conditionally
to a sip, isdn, or voicemail.
No google is involved as i use a local xmpp server (ejabberd)
I was experimenting on 1.8.15.1 (with jabber.conf,
Hi all,
In one of my other project i had a look at chan_mobile.
I build 1.8.15.1 with the apropiate module. (in my distro asterisk is
build without chan_mobile ;-)
After i filled in the mac-addresses of the BT-adapter and the one from
my phone, i see it is recognized, got connected, and
On Tue, 2012-09-18 at 17:43 +0100, Sebastian Arcus wrote:
Hi Hans,
The following page has some useful info:
http://www.voip-info.org/wiki/view/chan_mobile
Sebastian
Indeed. Didn't realise it was so picky.
just bought a couple of bt-adapters.
Will try again tomorrow and feed the
On Wed, 2012-09-12 at 00:01 -0500, Vladimir Mikhelson wrote:
Hans,
I did not try 10 or 11 as I run 1.8.15. Following are the related
conf files.
gtalk.conf
[General]
context = default
allowguest = yes ; Required if you want to accept calls
from people Not on your contact
Hi all,
Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and
11 version of asterisk.
In each example i got the impression that the asterisk server is
registering on a XMPP server as a single user with the credentials as
specified in jabber.conf.
Instead of a single xmpp-user,
On Tue, 2012-09-04 at 13:58 +0500, qasimak...@gmail.com wrote:
How about stripping it down to bare minimum's?
How about an other ARM-board?
http://gooseberry.atspace.co.uk/?page_id=13
Specifically the more mem (4GB) will help..
hw
--
Hi all,
After making a nice demo-setup for one of our innivationmanagers, he
came up with a completely different stratagy ;-(
They want to have an Ejabberd server, with xmpp-clients.
When you see a contact coming online, just point and click for making a
phone call.
Sounds/looks nice and
On Thu, 2012-08-23 at 15:01 +0530, Gopalakrishnan N wrote:
Hi,
Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1
(32bit) version in virtualbox. Downloaded Asterisk 1.8.15. Installed,
installation went fine.
Have you tried the versions from the OBS?
Or perhaps a
On Wed, 2012-08-01 at 19:39 +0800, D Tucny wrote:
snip
For reference... In my opinion HP servers should never be bought
without the battery or alternative, they shouldn't even be offered for
sale without it...
In my case, our purchase department changed our order.
They thought in their
On Sun, 2012-06-03 at 23:23 -0400, Tom Browning wrote:
Any tips on solving the following performance conundrum:
Asterisk 1.8.12.2 running on HP DL360 G5 and G7s
tcpdump running to capture UDP 5060/SIP signaling to .pcap files
All calls are ultimately B2BUA client - asterisk - PSTN
On Tue, 2012-07-24 at 11:07 +0530, Kannan wrote:
Hi Stelios,
Thanks for the response.
I take the following excerpt from your response. --- You can, but
usually for virtual/hosted pbx's you need an additional
layer of management software or a lot of copy paste
Could you please
On Wed, 2012-07-18 at 02:27 -0400, Jeremy Kister wrote:
I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10.
The system itself is happy and phone calls (between two parties) seem fine.
Unfortunately, when a caller listens to a Playback recording, there
seems to be
On Wed, 2012-07-04 at 10:15 +0530, Chandrakant Solanki wrote:
So, is http://sourceforge.net/projects/aterisk-amr/files/ same patch
also works in 1.8.13.0??
On Wed, Jul 4, 2012 at 3:18 AM, Hans Witvliet aster...@a-domani.nl
wrote:
On Tue, 2012-07-03 at 17:13 +0530, Chandrakant
On Tue, 2012-07-03 at 17:13 +0530, Chandrakant Solanki wrote:
Hi All,
OS : Cent OS 5 64Bit
Asterisk : 1.8.0-rc2
AMR Source Link : http://sourceforge.net/projects/aterisk-amr/files/
When I tried to call or start asterisk, I found Segmentation Fault.
Without trying to be pedantic, but
On Sun, 2012-07-01 at 13:04 +0530, alok srivastava wrote:
dear
i have configured properly asterisk. At the one end i am using x-lite
soft ph and another end twinkle. call is going properly from both end
but after picking the phone not able to listen other one.
when i checked the port 5060 on
Hi,
Couple of moments ago my asteriskbox with a bri-card went down.
(burn-out)
I've heard that it seems to be possible to use an fritz!box as an
isdn-gateway (isdn -- sip)
Anyone around who has good/bad experiences with those AVM-boxes?
(yeah, i know it is tech overkill, but i'll get an
On Thu, 2012-05-10 at 07:40 -0500, Tim Nelson wrote:
- Original Message -
On Thursday 10 May 2012, Bart Coninckx wrote:
I'm looking for a smaller,
appliance-type like PC, preferably solid state and fanless PC.
Since it's only going to run Asterisk for a couple of extensions I
On Mon, 2012-05-07 at 19:03 +0100, Roger Burton West wrote:
On Mon, May 07, 2012 at 12:03:17PM +0200, Bart Coninckx wrote:
What about phones like the Unidata WPU-7800 (
http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have
experience with those? Would these also suffer from
On Thu, 2012-03-08 at 16:50 +, Gavin Henry wrote:
Ah, this makes sense now. So as of today the status of TLS and SRTP in
anything
other than 1.4.X is unknown?
Umm... no :-)
OK, sorry :-)
Asterisk 1.4 did not have support for SRTP or SIP/TLS. Thus, neither of
these were
On Tue, 2012-01-31 at 15:52 -0500, John Knight wrote:
I like the idea of LTR release more often that would have the
feature patches baked in. Case in point the new conference app
requires a jump to version 10 while the 1.8 conference app is quite
useless but 1.8 is my LTR version so I am
On Fri, 2012-01-06 at 16:00 -0600, Tom Poe wrote:
Just installed asterisknow 1.6. I can access freepbx. I need to test
system on my LAN. Which softphone is best to use? I'm running ubuntu
on Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX
for incoming/outgoing
On Sat, 2011-12-03 at 15:36 -0500, Nick Khamis wrote:
Hello Everyone,
Are there any descent generic IVR recordings, that we can
use to quickly get our PBX up and running? It will obviously
not include the company name.
It's easy enough to make your own recordings.
Word of caution though.
On Mon, 2011-12-05 at 18:51 -0800, Steve Edwards wrote:
snip
Your security needs depends on your environment. At this point in time,
all of the hosts I manage for my clients exist in very limited
environments and have very small attack surfaces. They are racked in
secure data centers. They
On Wed, 2011-11-30 at 20:03 -0500, Adam Moffett wrote:
You can make a pretty good prediction with ping.
sudo ping -f -i .02 -s 180 -Q 0xb8 [ip] gives a tolerable simulation
of voip traffic. let it run for awhile, then press ctrl+c and see how
many packets were dropped and also check the
On Thu, 2011-12-01 at 14:02 +, A J Stiles wrote:
On Thursday 01 December 2011, gincantalupo wrote:
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
1. Migrate your Skype users over to a better product which supports proper
open standards.
On Thu, 2011-12-01 at 04:52 +0530, NaJIm wrote:
Is there anything else that I should be concerned about, when looking
to signup for a SIP provider. ??
Latency is important, but packet loss also, likewise packet re-ordering.
hw
--
On Wed, 2011-11-09 at 16:10 +0300, Anton Kvashenkin wrote:
Is anybody using pci-passthrough?
Yes, though quite a while ago.
About three years ago, i used pci-passthrough to give a dom-U access to
a localy mounted smartcard.
But i have a vague feeling that you are up to something else...
I know
On Mon, 2011-11-07 at 11:45 -0500, Nick Khamis wrote:
That sucks! What about KVM or XEN?
Nick.
No problems here with XEN.
(Perhaps i should mention, that i use paravirtualsisation to get the
best performance.
Distro: mix of SLES11sp1 /open_11.4)
hw
--
On Tue, 2011-11-01 at 12:08 -0500, Tim Nelson wrote:
Greetings-
I'm about to dive into the process of virtualizing some of my Asterisk
(primarily 1.4.x) infrastructure. In the past, when looking at virt
solutions, the primary issue preventing me from moving was the lack of proper
timing.
On Fri, 2011-10-14 at 10:02 +0300, Muro, Sam wrote:
Hi there
Consider this. You have three SIP extension 200, 201 and 202 and you have
configured your phones, say Polycom 331 to those accounts. 200 being one
very sensitive individual.
Lets say, an insider, get a new phone or perhaps an
On Wed, 2011-09-14 at 09:00 +0100, Ishfaq Malik wrote:
On Wed, 2011-09-14 at 00:44 +0200, Hans Witvliet wrote:
Hi all,
I presume i made a silly mistake while filling a database
But while googling on the results, i came across a lot of messages about
the layout of app_data
Hi all,
I presume i made a silly mistake while filling a database
But while googling on the results, i came across a lot of messages about
the layout of app_data in case of goto and dial statements.
(mostly about using the old | seperator instead of the , separator.
So i was wondering, is
On Fri, 2011-09-02 at 11:33 -0500, Kevin P. Fleming wrote:
On 09/01/2011 04:39 PM, Hans Witvliet wrote:
From the asterisk-bible and the wiki's i learned that it is possible to
let asterisk do some of the presense-info by means of the jabber.conf
file and a seperate xmpp-server.
You
Hi all,
Last couple of days i've arguing with my colleges about presence-info.
From the asterisk-bible and the wiki's i learned that it is possible to
let asterisk do some of the presense-info by means of the jabber.conf
file and a seperate xmpp-server.
On the other hand, most soft-phones are
On Thu, 2011-09-01 at 21:32 +0530, RSCL Mumbai wrote:
My main interest of being on Virtual platform is portability / Backup.
In case of any h/w issues, or crashes, simply copy the VM on to
another box and you are up in minutes.
Sanjay
--
Doing that right now, although in my
On Fri, 2011-08-26 at 19:03 -0400, Eric Wieling wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Friday, August 26, 2011 6:09 PM
To: Asterisk Users Mailing List - Non-Commercial
Hi all,
I know that a lot of people have negative experiences with
grandstream-2000, but personally. i'd only the repace one poweradapter
after three years...
So, can anybody give some comment on one of their recent models,
the GXV-3175 (the one with the 7 display)
I'm looking for a phone with
On Fri, 2011-08-26 at 07:41 -0700, Steve Edwards wrote:
On Fri, 26 Aug 2011, linux guy wrote:
How much power does the home asterisk box need ?
I use a small box (like those hp thin clients)
But these are a bit stronger aluminium housing, instead of plastic,
and better foor cooling.
Power
On Fri, 2011-07-22 at 10:58 -0700, Dave Platt wrote:
They've got a bunch of Grandstreams that seem to be rock solid... until
7:00pm. At 7:00, some of the phones become unavailable, and stay down.
Call
quality is solid almost all the time. But right at 7:00, things go bad.
Only
Hi all,
Perhaps a no-brainer, but i think i am making my dialplan on my proxy
too complicated.
Normally, what you find in the examples is that you have to dial a
specific number, other 9 or 0 for an external line.
What i want to do is this:
If you pre-pend a number with something like * then
Hi all,
Trying to find where i got wrong in my config
Is the realm parameter in sip.conf only used for possible
autentication?
The thing is, i got my box more-or-less working as i wanted,
but i can only reach internal functions (like echo-test and so on) and
other sip-clients if i dial
On Tue, 2011-07-05 at 12:33 +0500, Faisal Hanif wrote:
The problem you are reporting is not related to realm but can be context or
domain.
Tnx,
It was indeed a domain issue.
In some cases static definitions in /etc/hosts is not a good replacement
for DNS...
hw
--
On Fri, 2011-06-10 at 05:52 -0400, Steve Totaro wrote:
On Fri, Jun 10, 2011 at 1:48 AM, Hans Witvliet h...@a-domani.nl wrote:
On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote:
On Thu, 9 Jun 2011, Hans Witvliet wrote:
I went originally from a almost working machine running
Hi all,
I got three asterisk-machines, two of them acting as proxies.
On one machine (sles11sp1) i got iritating messages about not bing able
to find codec's and other stuff, so i thought it might be time for an
update: Stupid!
I went originally from a almost working machine running:
On Fri, 2011-06-10 at 07:21 +0800, Larry Moore wrote:
On 10/06/2011 5:32 AM, Hans Witvliet wrote:
Hi all,
I got three asterisk-machines, two of them acting as proxies.
On one machine (sles11sp1) i got iritating messages about not bing able
to find codec's and other stuff, so i thought
On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote:
On Thu, 9 Jun 2011, Hans Witvliet wrote:
I went originally from a almost working machine running:
asterisk180-1.8.3.2-87.1
To a machine that continuously restarts asterisk (+core dumps) running:
asterisk180-1.8.3-85.2
Any
Hi all,
I've got something strange, that got me searching for quite awhile.
Configuration as followed:
Linphone on a laptop, that is connected via openvpn to a proxy.
That proxy is connected with iax to another asterisk.
On the second one i have several hard and softphones.
Behaviour at first
On Fri, 2011-06-03 at 09:07 +0100, Ishfaq Malik wrote:
Are you suggesting that there are no bugs in 1.4 or 1.6?
I presume that you are aware of the fact that it is impossible to prove
the absence of bugs in any piece of software
You might not have detected them yet.
Furthermore behaviour
On Mon, 2011-05-30 at 23:15 -0400, Jeff LaCoursiere wrote:
On Mon, 30 May 2011, Sherwood McGowan wrote:
True, but with all due respect, if the cache's TTL expires and the OP's
PBX cannot reach an external DNS server, they have bigger problems ;-)
Slainte all!
The Mick
I
On Tue, 2011-05-31 at 10:29 -0400, Eric Wieling wrote:
As far as I can tell it is trying to do a reverse lookup on the IPs
configured on the system. With the internet down, does the command host
10.10.10.1 (or whatever IPs you have on the system) take a while to come
back? Unless you can
On Mon, 2011-05-30 at 13:57 +0530, virendra bhati wrote:
Thanks a lot all,
Now my view is clear ...
On Sun, May 29, 2011 at 3:15 PM, Gordon Henderson gordon
+aster...@drogon.net wrote:
On Sun, 29 May 2011, virendra bhati wrote:
Hi List,
On Fri, 2011-05-20 at 10:05 +0100, Andrew Thomas wrote:
Post your cdr_mysql.conf and res_mysql.conf and we'll take it from
there.
Don't forget to remove any 'private' info first (like passwords).
Cheers
Tnx for the offer,
Wil get the files when got back at the office.
I presume that
Ok, i tried the suggestion:
Instead of:
sippuser = resource, database_name, table_name
sippeer = resource, database_name, table_name
I put in:
sippuser = resource, context, table_name
sippeer = resource, context, table_name
Unfortunately, with the same results.
btw i tried both general as
Still a couple of questions..
I did configure extconfig.conf
...
;iaxusers = odbc,asterisk
;iaxpeers = odbc,asterisk
;sipusers = odbc,asterisk
sipusers = mysql,asterisk,sip_devices
sippeers = mysql,asterisk,sip_devices
;sippeers = odbc,asterisk
;sipregs = odbc,asterisk
;voicemail =
On Sat, 2011-05-14 at 19:51 -0400, Bruce B wrote:
Hi everyone,
I want to issue the command:
iptables -F
and then rebuild everything from the beginning with a very limited
scope and then without locking myself block all other traffic. Can you
suggest what I should put in the shell
On Thu, 2011-05-12 at 21:30 +0200, bakko wrote:
Hi,
look if you have res_config_mysql.so module instaled on your asterisk.
On CentOS /usr/lib/asterisk/modules
Regards
Tnx for your reply.
It turned out, that mysql-support was in a different rpm (addons)
As systems are never connected to
Hi all,
A week or so down the list, i read that not many people were using
realtime on an Asterisk18, so i had this afternoon a go at it...
[sorry for the inconveneant line-wraps]
First i did:
mysql create database asterisk;
mysql grant all on asterisk.* to 'voipadmin'@'localhost' identified by
On Sat, 2011-05-07 at 16:24 +0100, --[ UxBoD ]-- wrote:
I know a lot has changed over the past couple of years, and even
monthly, and that Asterisk running within a virtualised environment is
very happy indeed. If one would only be using SIP/IAX would Xen/KVM be
the best solution ? / or
On Wed, 2011-04-27 at 21:34 +0200, Olle E. Johansson wrote:
Friends,
We have a discussion on asterisk-dev about the maintenance of the 1.4 branch.
According to the release plans, support for 1.4 was scheduled to close in
April 2011 - basically now.
After that, only security patches would
1 - 100 of 285 matches
Mail list logo
