to itself, irrespective of
what the DB users did :-)
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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Be aware that there are several different conferencing solutions for asterisk.
I've used app_meetme in asterisk 1.8 (the LTS release) pretty happily. It is
reasonably full featured and
well supported. It has 2 drawbacks : 1) it needs a kernel module (Dahdi) to do
the mixing and
timing 2) it
I'm quite fond of GSM610 as a low(ish) bandwidth codec - although it isn't as
good as (say) speex or Silk,
it is widely supported, and European users have had years of cellphone use to
get used to the specific
sound of a GSM call. So you can often go from a GSM610 supporting handset all
the
, including g722 :-)
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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On 23 Jun 2011, at 13:44, randulo wrote:
On Thu, Jun 23, 2011 at 1:58 PM, Tim Panton t...@westhawk.co.uk wrote:
The good news is that it supports a load of nice codecs now, including g722
:-)
And you know what that means?
Unfortunately it means it doesn't work (yet).
You should
of the management are pilots too so it seemed like a
good idea. Says Mark Spencer whilst
dodging a cloud of his creations
T.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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language.
For example SIPp has its own framework, where If the asterisk device is
sending 100 message, SIPp is capable of recognizing that. In that way I am
asking.
On Fri, Apr 1, 2011 at 8:13 PM, Tim Panton t...@westhawk.co.uk wrote:
Gosh, it depends what you want to do with asterisk
}, // filter to just the events we care
about
{event - doSomethingZipDXSpecificHere( event.callerIdNum ) } // do
something
});
If I were to polish this up and make an open source framework of it would
anyone use it ?
Tim.
Tim Panton - Web/VoIP consultant and implementor
error
2) cable issue (I made them)
3) hardware problem with the Digium card
4) software (lib pri)
Any clues? Anyone seen this recently (google shows it in 2005 but not since as
far as I can see)
T.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
Asterisk 1.8.2.3 built by thp @ Asterisk01 on a x86_64 running Linux on
2011-02-07 14:32:26 UTC
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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easily.
Thanks for the help.
sean
Anyone know how iaxagent is accessing the speaker/mic ?
In theory the phone should have echo cancellation built-in, but it may only be
enabled in
certain cases.
T.
Tim Panton - Web/VoIP consultant and implementor
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I'm looking to connect a BMC 450 to an asterisk with a Digium Quad E1 card.
Am I right in thinking that I'll need a special 'crossover-E1' RJ45 cable?
If so, any clues where I might buy one in the UK? The Digium card sellers don't
seem to
stock such a thing.
Thanks.
Tim.
Tim Panton - Web
optimistic it will be fixed soon.
Tim.
Tim Panton - Web/VoIP consultant and implementor
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in case it isn't just me)
T.
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On 8 Oct 2010, at 15:37, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton
Sent: Friday, October 08, 2010 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On 8 Oct 2010, at 16:03, Bryant Zimmerman wrote:
Tim
I am actually seeing this on a 1.6.2.13 box as well. For some reason durring
prompt playbacks they some times stall mid file. The call stays up but no
audio comes in.
Bryant
From: Tim Panton t...@westhawk.co.uk
Sent: Friday
in iax.conf
I've never tried it, but I'd be happy to co-operate on an experiment :-)
Tim.
Tim Panton - Web/VoIP consultant and implementor
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?
(This is on asterisk 1.8 svn trunk - but I don't think that is important,
I think it is a package number issue)
Thanks in advance,
Tim.
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know if OpenBTS support handoff?
Thanks
On Fri, Aug 20, 2010 at 12:32 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Fri, Aug 20, 2010 at 10:41 AM, Tim Panton t...@westhawk.co.uk wrote:
On 19 Aug 2010, at 20:59, Randy R wrote:
On Thu, Aug 19, 2010 at 12:37 PM, Alan
On 23 Aug 2010, at 18:07, Warren Selby wrote:
On Mon, Aug 23, 2010 at 11:34 AM, Tim Panton t...@westhawk.co.uk wrote:
What is menuselect actually looking for when it blocks me from selecting
res_odbc ?
I've got unixOdbc installed and working. I also have /usr/lib64/libltdl.so.3
- so
,
although I'm not the radio guy - asterisk is more my thing :-)
Tim.
Tim Panton - Web/VoIP consultant and implementor
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New
On 29 Mar 2010, at 08:13, Tzafrir Cohen wrote:
On Sun, Mar 28, 2010 at 09:16:48PM -0500, Tim Panton wrote:
On 28 Mar 2010, at 10:13, Tzafrir Cohen wrote:
On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote:
I'm having trouble getting a xorcom set up.
A large part of the problem
, but I suspect it is loading the wrong version.
I got e4e4:1164 (I think - I've lost contact with the box for the moment).
Thanks for the explanation too.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
On 28 Mar 2010, at 10:13, Tzafrir Cohen wrote:
On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote:
I'm having trouble getting a xorcom set up.
A large part of the problem is that the box is a _long_ way away and
I can't get to/at it easily, so while I could probably fix
Any hints as to what I'm doing wrong would be much appreciated.
(here's some project background for the curious
http://babyis60.wordpress.com/2010/02/25/the-island-phone-system-adventure/ )
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
of 729)
3) downloading the soundfiles in 729 (you currently only have GSM)
Do 3) anyway - gsm transcoded to 729 always sounds horrible.
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
smime.p7s
Description: S/MIME cryptographic signature
Browser+quicktime: gsm,mp3,wav etc
Browser+flash: MP3 (and perhaps speex)
Browser+java: Pretty much any format you like
I wrote (and opensourced) a little java applet that plays .gsm files
see http://www.westhawk.co.uk/software/playGSM/PlayGSM.html
Tim.
Tim Panton - Web
a protocol violation
- for them to be sending a secret without a username.
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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Is there a version of this patch for 1.6.2 - or did the recent 1.6.2
rc1 drop include it ?
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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problems ?).
Tim.
Tim Panton - Web/VoIP consultant and implementor
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AstriCon 2009 - October 13 - 15
the encoding quality
(in codec.conf )
parameters to be different at the 2 ends. The speex decoder should at
the far end
should be fine with that.
see http://www.voip-info.org/wiki/view/Asterisk+config+codecs.conf
Tim.
Tim Panton - Web/VoIP consultant and implementor
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Register Now: http://www.astricon.net
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skypeforasterisk process running. I tried to killall
-9 asterisk but it did not solve my issue.
Any other suggestions?
Thanks for your help,
Emrah
Tim Panton wrote:
I had that too, I cured it by kill -9 'ing the skypeforasterisk
process that was left over from
the previous version of the beta.
Hope that helps
= 212555,1,Goto(companyb,${EXTEN},1)
; then separate contexts for each company:
[companya]
extern = 212555,1,.
extern = 212555,2,.
[companyb]
extern = 2125551112,1,.
extern = 2125551112,2,.
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
along about IAX
to a very senior Digium employee, which also proves nothing much :-)
Competition is a good thing - even amongst protocols.
T.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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a ghost call entry.
It would be interesting to know what state those ghost calls are in -
iax2 show netstats
on the CLI might tell you something interesting.
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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free to call in and try it this
Friday.
Tim.
Tim Panton - Web/VoIP consultant and implementor
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of 'data' transports, including the one
that was used
to upgrade the IAXy firmware.
I don't think you would have to change much (if anything) in the
protocol
to make it work.
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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Tim Panton - Web/VoIP consultant and implementor
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...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com
] Im Auftrag von Tim Panton
Gesendet: Samstag, 9. Mai 2009 11:46
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] Rusting Snoms?
This is a bit off topic, because I 'think' it isn't an Asterisk
Nachricht-
Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com
] Im Auftrag von Tim Panton
Gesendet: Samstag, 9. Mai 2009 11:46
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] Rusting Snoms?
This is a bit off
easily get on the edge - or
over the edge. If you have another switch/different model, a quick
try will help isolating the problem.
CS
-Ursprüngliche Nachricht-
Von: Tim Panton [mailto:t...@westhawk.co.uk]
Gesendet: Dienstag, 19. Mai 2009 13:46
An: Asterisk Users Mailing List - Non
' in those
(old) phones while they were switched off. Has anyone got any clues
for me?
Thanks!
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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64-bit RHEL rpm available anywhere?
Make sure you have the odbc headers installed (is that unix-odbc-
devel ?)
Then re-run ./configure before you do a make menuselsect
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
smime.p7s
Description: S/MIME cryptographic
, no
connection to
the company).
I'm running asterisk 1.4 on an NSLU2 , only a couple of channels
and minimal
transcoding, but it seems fine and stable. £80 + usb storage
I built 1.4 from sources on the NSLU2 which took a while :-)
T.
Tim Panton - Web/VoIP consultant and implementor
/call.xsql?key=echo123
(a quick demo I knocked up with the SFA beta)
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
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using the existing IAX libraries, I'd have thought,
however...
Gordon
It would be pretty easy to take the Mexuar Corraleta Java IAX source
and make a commandline Jar
from it. Such a jar would work on Linux, Mac and Windows.
T.
Tim Panton - Web/VoIP consultant and implementor
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support this.
What were you planning to do with it.
Tim.
On 16 Mar 2009, at 13:04, Giorgio Incantalupo wrote:
Hi Tim,
ok, but I think the big question is...what is the URL for? It seems I
need a special device...but which? What kind of device do you use?
Thanks.
Giorgio
Tim Panton wrote
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on the VoIP user
conference on friday
(I'm on a jittery hotel wifi so a bit garbled.)
http://recordings.talkshoe.com/TC-22622/TS-198841.mp3
Also briefly covered in my blog on ecomm :
http://tinyurl.com/b60-ecomm
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
smime.p7s
* in this way and it works like a charm. We can hit much
more than 2400 baud I think too.
--Dave
Our creditcard company's small print _insists_ on a direct analog
exchange line
with no other devices in between.
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
.
I will post it here as soon as i have the page up ...
If you plan to do significant work on it, please could you put it on
sourceforge
so others can chip in ? (That's kinda the point of GPLing it)
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
.
For personal reasons I'm not keen to be the project owner,
but I will contribute when I can.
Tim.
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I'm delighted to be able to say that as part of the agreement on my
departure from Mexuar,
the Corraleta applet source code Westhawk Ltd wrote for them has been
released under the GPL.
it is available for download at :
http://www.mexuar.com/files/corraleta_sdk.rar
Tim.
On 20 Sep 2007, at
Cognation Inc
d...@cognation.net
+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
+44-20-3129-6001 (London in-dial).
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Tim Panton
Sent: Wednesday, 14
On 14 Jan 2009, at 16:47, Matthew Rubenstein wrote:
Thank you for getting that code contributed to the community. Is
there
a spec somewhere of the features supported by that applet? A version
history? Docs of the SDK it's distributed as?
All I have is the link.
I should emphasise
On 14 Jan 2009, at 17:07, Roberto Fichera wrote:
Tim Panton ha scritto:
It isn't really in a state for novices at the present
you'd need:
1) a java compiler
2) a java code signing certificate (java applets can't read from the
mic
without being signed)
3
On 14 Jan 2009, at 18:02, Roberto Fichera wrote:
Tim Panton ha scritto:
On 14 Jan 2009, at 17:07, Roberto Fichera wrote:
Tim Panton ha scritto:
It isn't really in a state for novices at the present
you'd need:
1) a java compiler
2) a java code signing certificate (java applets
On 14 Jan 2009, at 18:11, Matthew Rubenstein wrote:
On Wed, 2009-01-14 at 17:38 +, Tim Panton wrote:
On 14 Jan 2009, at 16:47, Matthew Rubenstein wrote:
Thank you for getting that code contributed to the community. Is
there
a spec somewhere of the features supported by that applet
(and since it isn't in the .rar I guess they don't intend to at the
moment).
The easiest thing would be to run JavaDoc over the applet class and
see what public methods exist.
Tim.
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On 14 Jan 2009, at 18:36, Roberto Fichera wrote:
Tim Panton ha scritto:
On 14 Jan 2009, at 18:02, Roberto Fichera wrote:
Tim Panton ha scritto:
On 14 Jan 2009, at 17:07, Roberto Fichera wrote:
Tim Panton ha scritto:
It isn't really in a state for novices at the present
you'd need
One way to do this would be using
func_odbc.conf
This allows you to define dialplan functions that are based on ODBC
queries.
Like this, which looks up a meetme room number based on the project
and the 'space' number within that project (sub-project if you like).
[SPACE]
prefix=MEETME
If you set IAX2 debug on the HUNGARIAN machine and send the console
output
(or a wireshark output) I'll take a look.
At a guess it is a problem with your iax.conf file.
I generally find it clearer to have separate user and peer definitions
for
each system rather than relying on 'friend' which
On 1 Dec 2008, at 13:38, Giedrius Augys wrote:
2008/12/1 Tilghman Lesher [EMAIL PROTECTED]
On Monday 01 December 2008 06:15:15 Giedrius Augys wrote:
I'm working with asterisk 1.6. And I have success using
func_odbc with
one row query results (SELECT source,destination from cc
WHERE
I think it doesn't work across channel types.
So it works (if I recall correctly) in IAX or in SIP or in ZAP,
but not in mixture.
I think that if you have a Dial() that rings several extens,
then any of the technologies involved can pickup with *8
So if you have
Dial(IAX/fredSIP/billzap/mark)
On 22 Nov 2008, at 00:06, Michael Collins wrote:
Date: Fri, 21 Nov 2008 16:20:28 -0600
From: Terry Wilson [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Large Asterisk installarions (~10,
000
extensions), preferably at universities
To: Asterisk Users Mailing List - Non-Commercial
On 21 Nov 2008, at 21:12, Joseph wrote:
Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone.
http://moziax.mozdev.org/
I tried it yesterday on eee pc, connected to asterisk on local LAN
and the performance is terrible!
The delay is about 2sec or 3sec. and very bad echo.
I
Ok, I'll bite, what possible IAX bugs/shortcomings/features can cause
echo ?
Tim.
On 20 Nov 2008, at 18:47, Steve Totaro wrote:
Simple tests. Change from the highly touted IAX2 to SIP, but before
that, download X-Lite and see if you have the same delay. If you
don't then look at your
On 7 Nov 2008, at 08:49, Louis-David Mitterrand wrote:
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Louis-David Mitterrand wrote:
When monitoring an asterisk through its iax2 port I get these
warnings
at the console:
[Nov 6 13:15:15] WARNING[2209]:
On 7 Nov 2008, at 09:57, Louis-David Mitterrand wrote:
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote:
Your monitoring app is not sending valid IAX2 packets to the
server. If
it was sending a true IAX2 POKE, it would be a valid packet and
wouldn't
generate this warning
On 24 Oct 2008, at 17:00, Phil Knighton wrote:
Hello all
What I'm looking for is some plain speaking advice on ISDN.
Currently using 4 analog lines connecting via a four port TDM400P
FXO card. We need to physically move our installations, and on
requesting the analog lines be moved -
On 20 Oct 2008, at 20:01, Steve Anness wrote:
I am sure this has been discussed prior, however, I am sitting here
and being asked this very question by my superiors. They are loving
what I have done with our two Asterisk servers here; however, they
keep asking me if it is secure or
On 22 Oct 2008, at 07:23, Nikolai Lusan wrote:
On Mon, 2008-10-20 at 14:01 -0500, Steve Anness wrote:
However, realistically if I am using the asterisk server to make
internal calls and discussion very private matters, how possible is
it
for someone to listen to calls? How good is the
On 22 Oct 2008, at 10:44, voip crazy wrote:
Hello list,
Does anybody know any free WebCall solution to let our customer call
us directly via our web site?
Any clue will be welcomed.
Yep, take a look at our offering on www.phonefromhere.com
Tim.
On 22 Oct 2008, at 14:28, Rob Hillis wrote:
Tim Panton wrote:
Does anybody know any free WebCall solution to let our customer call
us directly via our web site?
Any clue will be welcomed.
Yep, take a look at our offering on www.phonefromhere.com
A per-minute charge does not constitute
Yep, we can probably help you, if you are interested send an email to
[EMAIL PROTECTED] and someone will get back to you to discuss
it.
Tim.
On 13 Oct 2008, at 18:58, Dean Collins wrote:
Tim Panton from Phone From Here was able to implement this
functionality when he was at Mexuar so I
On 26 Sep 2008, at 11:17, Grygoriy Dobrovolskyy wrote:
I have tryed skip2pbx 580€ yeastar 60 €, the quality is the way
behind of a good sip provider, thay are simply not suitable for
business, i hope it would not be the case of asterisk addon. Also i
wonder if skype auto relay will be
On 26 Sep 2008, at 04:36, Dean Collins wrote:
I'd also like to know what happens when someone 'chats' to the account
connected to the Asterisk server.
I asked Mark about that.
They expect to have text to work right, when associated with a voice
call.
It is less clear what happens it it is
It's essentially a channel driver.
Licensed per channel in the same way that the g729 codec is.
Limited private beta opening soon.
Tim.
On 25 Sep 2008, at 17:47, Steve Anness wrote:
So does this mean that my users who currently have skype running on
their
systems won't have to install
They demoed it - everyone seems pretty confident it works
as advertized.
No wide-band codec (yet)
Tim.
On 25 Sep 2008, at 17:55, randulo wrote:
I know a lot of linux and open source people think it's superfluous,
but a pseudo chan_skype is huge (assuming it works as advertised). It
means
On 17 Sep 2008, at 23:50, Philipp Kempgen wrote:
Tim Panton schrieb:
On 17 Sep 2008, at 14:57, Philipp Kempgen wrote:
Just a quick question
---cut---
[Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel
of type 'IAX2' (cause 34 - Circuit/channel congestion)
[Sep 17 15:52
On 17 Sep 2008, at 14:57, Philipp Kempgen wrote:
Just a quick question
---cut---
[Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel
of type 'IAX2' (cause 34 - Circuit/channel congestion)
[Sep 17 15:52:14] WARNING[8232] chan_iax2.c: No more space
[Sep 17 15:52:14]
On 12 Sep 2008, at 09:20, Michiel van Baak wrote:
On 09:59, Fri 12 Sep 08, Stephen Davies wrote:
xx-montague-gardens*CLI show uptime
System uptime: 38 years, 37 weeks, 4 days, 10 hours, 47 minutes, 11
seconds
Amazing. Especially considering:
[EMAIL PROTECTED]:/var/log uptime
On 9 Sep 2008, at 20:19, Mattias Andersson wrote:
Hi all!
I am looking for some software that would work as a proxy between a
SIP device (SIP phones and ATA boxes) and the Asterisk system
running IAX. The reason is that I can only get IAX trow the
firewalls, and can't change the
On 7 Sep 2008, at 21:34, Edgar Guadamuz wrote:
Hello,
I have been testing a trunk IAX and another SIP, using sipp to
generate SIP calls to a Asterisk box.
The testing dialplan just connects to another Asterisk box, who
answers the call and playback some files.
I noticed that the cpu
On 8 Sep 2008, at 13:12, Steve Totaro wrote:
On Sun, Sep 7, 2008 at 9:57 AM, Michiel van Baak
[EMAIL PROTECTED] wrote:
On 08:24, Sun 07 Sep 08, Steve Totaro wrote:
Maybe the problem is that IAX2 is not as set in stone as the RFCs for
SIP? Who is to say it is or isn't compliant to the
On 7 Sep 2008, at 08:38, Gordon Henderson wrote:
On Sat, 6 Sep 2008, hugolivude wrote:
OS = CentOS 5
Asterisk = 1.4.21
Router = WhiteRussian 0.9
Not sure whether I have a problem w/ Asterisk or White Russian
config,
so I'm posting to both lists.
I have 2 Asterisk servers running
I think I've forgotten something obvious
I've got 2 incoming calls, I want to bridge them - how can I do this ?
(assume I somehow know which calls should be paired up...)
I could dump them both in a meetme - but that seems wasteful
as i _know_ there will only ever be 2 parties. (And I need
I knew I'd forgotten something.
Doh!
On 5 Sep 2008, at 14:57, Andreas Brodmann wrote:
Tim,
you may want to try:
1) Park call 1
2) Pickup call 1 with call 2 (using ParkedCall)
Regards,
Andreas
2008/9/5 Tim Panton [EMAIL PROTECTED]
I think I've forgotten something obvious
I've got 2
On 5 Sep 2008, at 15:50, Steve Murphy wrote:
On Fri, 2008-09-05 at 12:27 +0100, Tim Panton wrote:
I think I've forgotten something obvious
I've got 2 incoming calls, I want to bridge them - how can I do
this ?
(assume I somehow know which calls should be paired up...)
I could dump
On 1 Sep 2008, at 17:34, Rob Hillis wrote:
VoIP Cyprus wrote:
Can you share with me your experiences with Asterisk 1.6? Is it
stable
enough for commercial service?
No. No matter how good some people may tell you it is, 1.6 is still
beta software and software is rarely beta for no good
On 31 Aug 2008, at 01:15, David Burgess wrote:
Asterisk Users -
We are presently try to operate a hybrid GSM/Asterisk cellular
basestation at the Burning Man Festival in the Nevada desert. (See
http://openbts.sourceforge.net). The architecture is basically one
where cell phones are
On 22 Aug 2008, at 14:55, randulo wrote:
On Fri, Aug 22, 2008 at 4:23 AM, Johansson Olle E [EMAIL PROTECTED]
wrote:
Just some friendly advice if you really want a discussion. Of course,
I clicked, read and commented ;-)
If this is a way we can get you to say something, Olle, I'm for it!
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 into
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
custom
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