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
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-) band
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.
Fi
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 p
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 lo
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
-Original Message-
From: Gergo Csibra
Reply-to: Gergo Csibra , Asterisk Users Mailing List -
Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ISDN outgoing caller id
Date: Tue, 27 Aug 2013 21:28:36 +0200
Tuesday, August
-Original Message-
From: Rafael dos Santos Saraiva
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Performance Asterisk large installation on
Vmware/Xen
Date: Sat, 18 May 2013 15:01:06
From: virus.c...@mail.ru
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
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-
exten => 911,1,Answer()
-Original Message-
From: jg
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] looking for a way to do appointment
reminders
Date: Fri, 26 Apr 2013 09:33:42 +0200
Hi Brandon!
I have
Might have a look at tine:
http://www.tine20.org/wiki/index.php/Admins/Asterisk_integration
hw
-Original Message-
From: Steve Totaro
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-user
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 R
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=::", an
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
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP account registration fails after
upgrade to 1.8
Date: Fri, 22 Mar 2013 02:46:43 + (UTC)
On Thu, 21 Mar 2013 16:
-Original Message-
From: Emiliano Vazquez
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] chan_mobile
Date: Tue, 12 Mar 2013 18:01:34 -0300
El 10/03/13 13:18, Hans Witvliet escribió:
> Hi,
>
>
-Original Message-
From: bilal ghayyad
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] digium card and virualbox
Date: Sun, 10 Mar 2013 20:18:52 -0700 (PDT)
I am not mixing. I need this for LAB testing.
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 ast
-Original Message-
From: Carlos Alvarez
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk with 1000 extensions
Date: Thu, 7 Mar 2013 09:30:31 -0700
On Thu, Mar 7, 2013 at 1
-Original Message-
From: termo termosel
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Error to install Asterisk
Date: Tue, 5 Mar 2013 14:30:05 +
Hi,
if I write du -sh the response is 271M. I don'
e mid 90's, but not 4G
John Novack
Hans Witvliet wrote:
> Are these 4G comaptible
>
>
> -Original Message-
> From: Frank
> Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
>
> To: Asterisk Users Mailing List - Non-Commercial Discuss
Are these 4G comaptible
-Original Message-
From: Frank
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] GSM Sip Gateway
Date: Sun, 24 Feb 2013 07:40:19 -0500
USA, this will
-Original Message-
From: A J Stiles
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk SMS()
Date: Tue, 19 Feb 2013 16:50:10 +
On Tuesday 19 February 2013, Nicholas John
-Original Message-
From: Carlos Alvarez
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk calls between 2 private networks
Date: Thu, 7 Feb 2013 10:36:36 -0700
On Thu, Feb
-Original Message-
From: Olivier
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote
LAN workstation
Date: Thu, 31 Jan 2013 08:25:42 +0100
Hel
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 m
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, a
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.
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 ex
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 [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 asterisk (and
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 o
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
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) rai
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, whe
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
h263)
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 Republi
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' fil
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, jin
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 th
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 immediate
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 yo
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, c
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 do-able
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
On Wed, 2012-08-01 at 19:39 +0800, D Tucny wrote:
>
> 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 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
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 -> PST
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
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
> wrote:
> On Tue, 2012-07-03 at 17:13 +0530, Chandra
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 pedant
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
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 dualb
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 ext
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 c
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, nei
On Mon, 2012-02-13 at 09:32 +0100, Benny Amorsen wrote:
> "Jason W. Parks" writes:
>
> > I can move my voice infrastructure to an IP-based one running 10Mbps,
> > utilize existing wiring infrastructure, with the only cost outlay
> > being low cost PoE managed switches (48 ports for about a grand)
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 s
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 c
On Mon, 2011-12-05 at 18:51 -0800, Steve Edwards wrote:
> 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 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 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
> op
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 chec
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 kno
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
> ti
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
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
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, i
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
> >
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
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 c
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 v
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: Ast
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.
Powe
On Wed, 2011-07-27 at 09:44 -0400, Claude Hayn wrote:
> We are frequently losing power during lightning storms. (Yes we have
> UPS, but often by the time power comes back up the UPS has run out of
> juice)
>
>
> Does anyone know of a solution for this issue? Having to get up in
> the late nigh
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.
>
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 "*" t
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
--
__
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 "12
On Fri, 2011-06-10 at 05:52 -0400, Steve Totaro wrote:
> On Fri, Jun 10, 2011 at 1:48 AM, Hans Witvliet wrote:
> > On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote:
> >> On Thu, 9 Jun 2011, Hans Witvliet wrote:
> >>
> >> > I went originall
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) r
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
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:
asterisk180-
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 gla
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 t
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
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
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 +aster...@drogon.net> wrote:
> On Sun, 29 May 2011, virendra bhati wrote:
>
> Hi List,
>
risk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans
> Witvliet
> Sent: 23 May 2011 13:42
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] [Fwd: FW: extconfig.conf]
>
> Hi Andrew,
>
> OK, (the simple fact th
Hi Andrew,
OK, (the simple fact that those machines are not connected to internet
makes that i have to go to those machines and copy them on a usb-stick,
so it causes some delay each time...)
Forwarded Message
Sorry - I meant extconfig.conf - not cdr_mysql.conf (my mistake).
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" a
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 => odb
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 i
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