On 21 Aug 2008, at 18:44, Jay R. Ashworth wrote:
> On Thu, Aug 21, 2008 at 09:40:04AM -0700, Michael Collins wrote:
>> To those running call centers I have a question: what kinds of
>> soft phones,
>> if any, do you use? I’m wondering what is out there that has
>> some hooks for
>> cu
On 20 Aug 2008, at 18:00, Eric Chamberlain wrote:
> We are exploring using Asterisk for a project and we are looking for a
> way to encrypt/decrypt the peer passwords stored in the realtime
> database (postrges).
>
> Ideally, we want to use a public key to encrypt the passwords before
> they go i
Erm, I've been out of the loop, but in 1.6 there's
the IAXVAR dialplan function that does _exactly_ what you want.
I don't know if it's been backported to 1.4, but I think there was a
patch
at one point.
Tim.
On 11 Aug 2008, at 20:43, Richard Lyman wrote:
> TP'n to follow broken flow.
>
> As
On 2 Jul 2008, at 18:16, Steve Edwards wrote:
> I'm fighting a losing battle with voiptalk.org in the UK.
>
>
> Now I learn that they don't support registration!
>
> They say "IAX will not work with our service without a public IP
> address
> or FQDN." (I think he meant to say "Our service re
Oops, you are right, has the fix.
(Note to self - "do try and pay attention and keep up!")
Sorry, ignore my previous post.
Tim.
On 2 Jul 2008, at 07:17, Michael J. Liberatore wrote:
> Are you sure your using 1.4.21.1 and not 1.4.21? I am pretty sure the
> "major bug" they fixed in .1 was the
On 1 Jul 2008, at 20:30, bilal ghayyad wrote:
Hi All;
I used Asterisk 1.4.21.1 and I discovered the following bugs, I do
not know if other used it and discover it:
1) In the IAX trunk, it suddenly stop working and I have to restart
the machine.
2) An FXS station, suddenly loose the ton
On 28 Jun 2008, at 18:36, Tzafrir Cohen wrote:
> On Sat, Jun 28, 2008 at 12:16:53PM -0400, Steve Totaro wrote:
>> On Sat, Jun 28, 2008 at 12:06 PM, Tzafrir Cohen
>> <[EMAIL PROTECTED]> wrote:
>>> On Sat, Jun 28, 2008 at 11:25:44AM -0400, Steve Totaro wrote:
On Sat, Jun 28, 2008 at 10:11 AM,
On 24 Jun 2008, at 13:27, Tilghman Lesher wrote:
On Tuesday 24 June 2008 06:44:21 Tim Panton wrote:
Is anyone using 1.4.21 with cdr_odbc and IAX channels successfully?
I'm getting lockups where asterisk stops responding (to anything).
Foolishly I've built a box with 2 new th
On 24 Jun 2008, at 13:09, Michiel van Baak wrote:
> On 12:44, Tue 24 Jun 08, Tim Panton wrote:
>> Is anyone using 1.4.21 with cdr_odbc and IAX channels successfully?
>>
>> I'm getting lockups where asterisk stops responding (to anything).
>>
>> Foolishly I&
Is anyone using 1.4.21 with cdr_odbc and IAX channels successfully?
I'm getting lockups where asterisk stops responding (to anything).
Foolishly I've built a box with 2 new things on it, 1.4.21 and Oracle
as
the odbc server.
If others are running it fine against MySql or postgres , I'll focus
On 24 Jun 2008, at 09:29, voip crazy wrote:
> Hello all,
>
> Someone knows any softphone which accept messages using sipsak?
> I just tried X-Lite and portsip without success
>
> Thanks
>
> Voipcrazy.
>
Take a look at firefly
http://www.freshtel.net/download/internetphone/
I'm pretty sure it doe
On 4 Jun 2008, at 21:00, Hilary Miller wrote:
> EdPimentl <[EMAIL PROTECTED]> wrote:
>> Have you seen these client?
>> http://www.mozillavoip.com/
>> http://tringme.com/
>> http://www.twoiplink.com/
>> http://www.openwengo.org/index.php/openwengo/public/homePage/openwengo/public/projectsFirefox
>
On 16 May 2008, at 00:26, Julian Lyndon-Smith wrote:
> I have a lot of recordings from asterisk in a .gsm format. I would
> like
> to play these files from a web browser (IE, firefox and opera)
>
> What do I need to do in order to achieve this goal ?
>
Sorry to catch up late on this, but I hav
On 8 May 2008, at 11:00, Adrian Marsh wrote:
Hi All,
Whats the SLN file format (for import/export to Audacity)?
Need to avoid Sox if I can
16 bit signed audio at 8khz.
2 bytes per sample, no compression, 8000 samples per second, network
byte order, no header.
Tim.
___
On 8 May 2008, at 09:36, randulo wrote:
> On Thu, May 8, 2008 at 5:40 AM, Steve Totaro
> <[EMAIL PROTECTED]> wrote:
>> Can this thread be moved to the biz list? It really does not belong
>> here when words such as "the best way to monetize an application or
>
> The topic is still salient IMO, bu
On 21 Apr 2008, at 14:31, equis software wrote:
> Hi!
> I need to implement click-to-talk web application.(not click-to-call
> or callback)
> I try to use njiax, and iaxclient but I can´t made it work.
>
> Has anybody other solution??
Yep. We can help on a commercial basis. Contact me off-list
On 8 Feb 2008, at 00:39, David Hogan wrote:
>> Alternatively you could fix the client :-)
>
> Heh :) Although it's a situation that won't happen in (our)
> production,
> for the sake of completeness I'll probably upgrade the client.
Actually, it does (assuming you guys still run Tesco's UK ser
On 7 Feb 2008, at 00:36, Tilghman Lesher wrote:
> On Tuesday 05 February 2008 09:22:29 Cavalera Claudio Luigi wrote:
>> Hello,
>> I'm doing some research concerning iax encryption, I haven't find any
>> clients (softphones or hardphones) which implement so I have not
>> tested
>> it yet.
>>
>>
On 7 Feb 2008, at 10:29, Vincent wrote:
> On Wed, 6 Feb 2008 20:12:21 +0100, randulo <[EMAIL PROTECTED]>
> wrote:
>> http://food4wine.ning.com/
>
> BTW, we also want to receive call notifications on our cell phones. In
> addition to using SMS, we found a cheaper alternative which is to use
> iMod
On 7 Feb 2008, at 09:17, David Hogan wrote:
> Hi all,
>
> I have spent some time searching, but I haven’t found a way to
> prevent * from concatenating two frames into one IAX packet.
>
> I have a situation where I make an IAX GSM call to *, which
> transcodes to an iLBC SIP call. Every secon
On 29 Jan 2008, at 11:08, George Pajari wrote:
> Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P
> (same problem with various previous versions; same problem with
> different TE120P cards).
>
> The customer has a partial (10 B-Channel) PRI that when it is busy
> (eight or more
>>
> Thanks for the replies.
> I wonder if I could use the Yealink phone and write a connector to
> Asterisk with the IAX client on Sourceforge and make the handset look
> like an iaxphone? Or maybe there is some other easier solution?
> All I
> need is to have the ability to go
> off hook/on h
On 8 Jan 2008, at 08:17, Armin Schindler wrote:
> On Tue, 8 Jan 2008, CSB wrote:
>> We are experiencing slightly distorted audio with playing of
>> recordings on
>> our Asterisk server when the call comes in over our Eicon Diva
>> Server BRI
>> card. An example is an incoming call to IVR and
The official URI for this is
tel:
see http://www.ietf.org/rfc/rfc3966.txt
It isn't implemented everywhere, but most cellphone browsers seem to.
Tim
On 4 Jan 2008, at 12:48, Dean Collins wrote:
> Snapanumber does this but with only certain browsers. (it doesn't
> work with ie which is what I
to trust that firewalls operate as
> they should - I've been bitten far too many times by a firewall that
> doesn't quite behave as you expect. Also, when diagnosing network
> connectivity problems, I find that it helps to have the rules in
> place rather than hav
30 calls in a trunk will be fine for IAX.
In fact IAX has a 'trunked' mode that could enable
that allows you to save
quite a lot of bandwidth by shrinking the packet headers
between a pair of asterisk systems.
Tim.
On 2 Jan 2008, at 15:26, bilal ghayyad wrote:
> Hi Rob;
>
> Big thanks for you
If you are careful, you only need to setup a port forward at one end
of the IAX trunk.
Have one Asterisk register (regularly) with the other.
The second asterisk (server) will need to have port 4569 forwarded
through it's router.
The first asterisk (client) wont need any port forwarding.
Tim.
On 2 Jan 2008, at 05:17, Vincent wrote:
> On Tue, 01 Jan 2008 16:10:47 +0100, MatsK <[EMAIL PROTECTED]> wrote:
>> The codec is specified (for a sip device) in sip.conf, like this:
>
> Good to know. Actually, I'll have Asterisk save voicemails as WAV and
> move the files to the www's htdocs, and s
On 3 Oct 2007, at 10:16, Mark Quitoriano wrote:
>
>
> On 10/3/07, Tilghman Lesher <[EMAIL PROTECTED]>
> wrote: On Tuesday 02 October 2007 16:55:52 Brian West wrote:
> > On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote:
> > > anyway still if there's a hack for meetme to work with g729 codec
> >
On 20 Sep 2007, at 19:46, Jean-Denis Girard wrote:
> Matthew Rubenstein a écrit :
>> Does anyone know of an IAX softphone in Java, whether
>> applet or
>> application? Even the most minimum featureset, just voice and
>> dialing,
>> or even embedded in some other app/let. Preferably GP
You can configure Mexuar's Corraleta web-based softphone to autoanswer.
The user won't be able to unconfigure this, except by quitting the phone
(or page or browser)
(Full disclosure - I'm a director at Mexuar and wrote much of the
code ;-) )
Tim.
On 17 Sep 2007, at 16:11, Joao Pereira wrote:
On 18 Sep 2007, at 18:11, Eric ManxPower Wieling wrote:
> If the #AsteriskNOW channel is dead on IRC that does not mean you can
> bring your problems to a channel dedicated to Asterisk (i.e. no GUI).
> Go ahead and use AsteriskNOW, but don't pester the people in
> #asterisk,
> most of which hav
On 14 Sep 2007, at 21:53, Ira wrote:
> At 01:11 PM 9/14/2007, you wrote:
>> Unfortunately, that seems to be more and more the way of the world,
>> though I will say that this kind of unproductive attitude and
>> intolerance of others is more prevalent on these kinds of computer
>> related lists t
On 14 Sep 2007, at 13:45, William Stillwell (Ki4swy) wrote:
> I am trying to determine what would need to be done/modified to
> enable the following:
>
> I have a SIP extension come into my asterisk box, and I then need
> it to call "6-10" remote Sip Stations that are set to Auto-Answer...
>
On 11 Sep 2007, at 12:32, Gordon Henderson wrote:
> On Tue, 11 Sep 2007, Juan Sandro wrote:
>
>>
>> Hi
>>
>> We have a number offices accommodating 4-6 people each hence it is
>> very
>> important for PBX to be fanless and silent. We have been looking
>> at using
>> IDE flash disks also calle
On 7 Sep 2007, at 17:56, phananhvu wrote:
> I means i want to use a software library to write a program that
> register an extension to Asterisk system. After that, i can bind my
> IP Phone to that extension.
> I wonder if Asterisk-Java can deal with this ??
Ah, you mean create an extension
address is, so that Asterisk knows where to send calls destined for
> that
> device. That's all there is to it.
For IAX it also has the side benefit of setting up path through nat
and port
mapping routers.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
__
'_.' => 7. NoOp("Here I am Mr Mgoo") [pbx_config]
> 8. Playback(tt-weasels) [pbx_config]
> Include =>'sip_autoreg' [pbx_config]
> -= 1 extension (2 priorities) in 1 cont
ifference.
Also I think 'iax2 show channels'
displays an (E) next to the channel if encryption is on.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
> I don't think creating a network without a single point of failure
> is unreasonable.
It's impossible. I can't think of a single example where this
actually exists.
Getting even close is hideously expensive.
Tim, speaking for himself :-)
application
> to use this API. I would prefer to use the SIP protocol, since it
> seems like its the most common.
We've got a commercial IAX based java softphone SDK that might help you.
Take a look at :
http://www.mexuar.com/products_sdk.shtml
Tim Panton
www.mexuar.net
www.
the voice frame within
> the trunked packet is encrypted. Any assistance would be appreciated.
I thought that Encryption and Trunking are mutually exclusive in IAX.
What does the iax debug in asterisk show?
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
___
On 17 Jul 2007, at 11:26, Steve Kennedy wrote:
> On Tue, Jul 17, 2007 at 11:56:35AM +0200, Anselm Martin Hoffmeister
> wrote:
>
>> Am Donnerstag, den 12.07.2007, 16:57 -0700 schrieb Russ McBride:
>>> Newbie question(s):
>>> From what I can determine it sounds like the SMS messaging isn't as
>>
; patch that does exactly that.
It's also in the svn trunk
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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Hmm, time to get that IAX encryption working along wit ZRTP
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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re sending via IAX or SIP. We had a problem where an internal id was
not
getting overwritten with a valid PSTN number, one of our suppliers
set a default caller-id and another rejected the calls.
The process is annoying, but it works fi
What does the cdr table you created in oracle look like ?
Tim.
On 20 Jun 2007, at 13:37, Everton Goularth wrote:
> Hi All,
>
> Thank's for your hint Tim Panton
>
> I could connect my asterisk machine to my oracle machine.
>
> I used unixODBC-2.2.11.tar.gz,
> oracle-i
;s meeting.
However, I'm curious to know which problems you
have that will be solved by V6.
Tim
Tim Panton
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www.westhawk.co.uk/
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fferent port. If they are blocking it by content inspection
(unlikely but possible I suppose) you could try turning on encryption.
Personally I'd try doing both the above before going down the VPN route.
However I know folks have VPNs working
your oracle system to check for an active connection
to 1521 from your asterisk.
- you should be good to go.
I've only used it for realtime iaxusers not for anything else yet. I
also have no clue how robust it is,
but I'll no-doubt find out soo
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On 12 Jun 2007, at 17:53, Rob Schall wrote:
We are trying to use a softphone from a location that is behind a
firewall. We are using x-lite as the softphone.
Alternatively you could use an IAX softphone. They generally don't
have a problem with NAT or firewalls.
Tim P
On 10 Jun 2007, at 13:29, Lenz wrote:
Hi Lee,
we are a Java shop and our experience with Java has been much the
one you say - it does scale pretty well and it is very solid. What
I was trying to say is that Java is not very well suited to the
classic, Unix-style, fire-up-process-and-let-
here else that might cause
these blips.
But the 2% CPU usage seems to suggest it shouldn't.
try running asterisk with the option -p
That might be a bit memory light I'd put 1Gb in that box.
Take a look at vmstat and see if it is
ry to restart the asterisk my agents not able to hear any
voice.
It looks like you are trying to get too many calls down a single trunk.
Try defining a few trunks and split your calls over them.
There is a limit on the total packet size in a IAX trunk.
Tim.
Tim Panton
www.mexua
tty
much useless,
the buttons are *TINY*, the battery life horrible, and the ringtones
gimmicky.
I haven't tried WEP or WPA on these things, but the phones I've
gotten rid of
long ago due to their problems.
-A.
If you can face the configuration hell, the Nokia e60 should do what
es from distribution to
distribution.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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solutions
alongside the pure voip style.
In our 'Corraleta connect' hosted solution we do exactly that.
There are some demographics where this isn't necessary,
it is up to the solution provider to identify this requirement - not
us as a technology provider.
Tim Panton
www.me
r things going on that
probably made it harder than it had to be.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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the party at the far end ?
Tim Panton
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en you get this message ?
Tim Panton
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www.westhawk.co.uk/
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capitalization error.
The font the nokia uses for it's config dialogs is such that the caps
don't stand out.
This was compounded by the fact that the default input method
capitalizes the first letter of every line.
I guess this isn't your problem, but if it is I've
idth
overhead, but it is
probably less that that of a VPN.
Note, the calling/called numbers are still passed in the clear over
encrypted IAX,
so you are still vulnerable to traffic analysis.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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work, then gave up
and had a perl script written that regularly posts the new CDR
records to oracle over http(s).
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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starting a project now, I'd
take a look at the (newish) support for IMAP storage for voicemail.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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f old ones from netgear
that have been pretty good.
The only drawback is that they only support a single device on the
ethernet side, so you'd need
one per phone.
Me, I'd go all modern on them and supply nokia e60's instead of the
p
produce better audio quality than asterisk
as it supports a wideband codec
but in practice asterisk/iax should be just as good and somewhat
more predictable as Skype's routing
varies from call to call.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
__
l you would need to do
is ensure the main asterisk powered itself off if it crashed, thus
dropping the
PRI.
Tim Panton
(speaking as westhawk)
www.westhawk.co.uk/
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rtunately I can't really mandate a mini-asterisk server and
Soekris box for each remote user - it'd add many hundreds of pounds
to the cost :-)
Hardly - Talk to [EMAIL PROTECTED] , he has a build that runs on
stock linksys routers, or
alternatively try an Nslu2.
The tric
asterisk supports this. So in this
case you are better off with a simple
codec like {au}law.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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On 23 Apr 2007, at 21:04, James FitzGibbon wrote:
On 4/20/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
I am not. The soft phone is not the only software on that computer
that
needs cetral configuration.
How do you configure the networking on those computers? The mail
clients? How do you de
at least 6 independent implementations of IAX I know of.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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kills Java's stats.
The paper I mentioned is more relevant to long running Voip programs
http://research.sun.com/techrep/2002/smli_tr-2002-114.pdf
Anyhow, lets take this off list if you want to discuss it further.
Tim Panton
www.mexuar.net
www.west
On 21 Apr 2007, at 13:06, Philipp Kempgen wrote:
Tim Panton wrote:
On 21 Apr 2007, at 03:21, Philipp Kempgen wrote:
Tzafrir Cohen wrote:
On Fri, Apr 20, 2007 at 11:48:20AM -0400, James FitzGibbon wrote:
Has anyone found a softphone that supports pulling it's
configuration from a
ce
usable (so most of them fail)
and should work on a real OS (tm). And no Java please :)
What's your objection to a softphone in java ?
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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www.westhawk.co.uk/
_
y
gently explains
that it is 6 weeks from order being accepted, and they haven't
accepted mine yet!
When are you going to accept it ? - About 5 weeks from when we plan
to fit it!
Hey, at least he was an honest bloke in a twisted system.
Tim Panton
www.mexuar.net
www
how/where your handset is connected. v6 helps, but not much.
Tim Panton
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cumbersome?
You could query via SNMP. it has the astChanVariables for each active
channel
as a DisplayString
I can't promise that this is less cumbersome, but the overhead might
be smaller.
Tim Panton
www.mexuar.net
www
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Voice frame - the far end will
retry after a while
and if that arrives all will be well from then on.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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or later, anti-trust
regulation will kick in.
Fun world.
If I were Vonage I'd have a delegation in HongKong now, moving all my
Telco interconnects
to somewhere where the US patent system is treated with the contempt
it is starting to earn.
Tim Panton
www.mexu
On 6 Apr 2007, at 21:21, Matthew Rubenstein wrote:
On Fri, 2007-04-06 at 12:00 -0700,
[EMAIL PROTECTED] wrote:
Date: Fri, 6 Apr 2007 16:13:29 +0100
From: Tim Panton <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Open Source VoIP client (on a webpage)
To: Jason Wolfe <[EMAIL
e is finding an easily
integrated open source client.
Any suggestions from those who know?
Our SDK isn't open source, but it is an IAX applet - javascript/DHTML
friendly and lightweight.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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daunting at first. It's a delight to have your
cell phone
be your officephone the moment you step into the wifi pool :-)
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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o increase the number of digits we sent!
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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u dig around on
http://www.voip-info.org/
You will find a long list with a (very) few entries that might meet
your requirements.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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uld hope that it could be implemented in Asterisk without too much
trouble.
Does the current work on SRTP extend into ZRTP?
At Etel I heard Phil Zimmermann say that he had a working implementation
of ZRTP for asterisk in the lab.
What was less clear was how/when this might be released.
T
some softphones) Is that what you want ?
Tim Panton
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www.westhawk.co.uk/
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alaw
packets instead of 160bytes.
Many endpoints seem to be ok with this, but corraleta for one doesn't
like this one little bit :-(
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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On 22 Mar 2007, at 08:30, Tomislav Parcina wrote:
Tim Panton wrote:
I once spent a week struggling with this sort of symptom to
find in the end that the ops guys had got fed up with my
line being in 'alarm' on their console and disabled it at their end.
One phone call later it was
On 21 Mar 2007, at 13:47, Matija Turk wrote:
I have one 120 channels isdn pri digium card. I worked fine for the
last 3 months. There was a power outage, and after the server came up
and isdn modem from the telecom, zaptel can't detect connection
(alarms are red). Currently I can't test with an
IAX2/0100101-3'
*CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX
Subclass: REGREQ
Timestamp: 6ms SCall: 6 DCall: 0 [192.168.52.95:4569]
USERNAME: 0100201
REFRESH : 60
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX
oil. So they have a completely silent machine
in 40C warm oil. Amazing...
I had a friend who filled her calculator with warm molasses, but that
was an accident :-)
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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ereal - or
with iax2 debg, send it to me with your iax.conf and I'll take a look.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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On 19 Mar 2007, at 14:31, Philipp Kempgen wrote:
Tim Panton wrote:
I really like the Mac Minis as small office servers, quiet, cool,
real UNIX, asterisk works on them.
The only downside is that you can't add PSTN cards.
Aren't there USB adapters available?
Anything needin
ot easy to find.
I really like the Mac Minis as small office servers, quiet, cool,
real UNIX, asterisk works on them.
The only downside is that you can't add PSTN cards.
Tim.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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911 or internal calls). Most folks are comfortable with
follow-me or redirection concepts.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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