work almost all UNIXs)
(b) is the more generic and preferred method IMO - it should work just about
everywhere... unless you have total disk encryption or encrypted filesystems
and are unable to mount the partitions... in which case... best of luck to
you.
--
Matt Watson
http://www.voipphreak.ca/2007/04/16/monitoring-asterisk-14-with-snmp-and-cac
ti-for-pretty-graphs/
Thanks,
Matt G
: <http://www.voipphreak.ca> http://www.voipphreak.ca
: <http://www.ratemydialplan.com> http://www.ratemydialplan.com
: <http://www.asterisk-jobs.com> h
On 12/28/08, jonathan augenstine wrote:
> Matt,
>
> Asterisk version == 1.4.22
> dtmfmode == info
> calls are bridged through Asterisk (canreinvite=no)
>
> Jonathan
Have you tried setting dtmfmode to 'inband' for both SIP endpoints?
MATT---
> On Sun, Dec
; would like the DTMF to pass-through to the other SIP phone. Is this a
> configuration issue? Or do I need to handle this on the dial plan level?
>
> Jonathan
Asterisk version?
What are both dtmfmodes set to for each SIP endpoint?
Are the calls natively bridged o
/dahdi.makeopts - but I have not verified
that.
--
Matt Watson
On Thu, Dec 18, 2008 at 11:49 AM, Jerry Geis wrote:
> >
> > Jerry Geis schrieb:
> > >/ Is there a way to install DAHDI and NOT download the echo canceler
> files?
> > />/ I dont have firewall acces
g... but i would expect very poor results.
3. You are right, you can';t really just make one yourself from scratch, you
need a source that has already been tuned properly to use as a reference for
creating your own.
--
Matt Watson
On Thu, Dec 11, 2008 at 11:01 AM, Olivier <[EMAIL PROTEC
> So, in short, if all my calls were from outside to a G729 enabled phone and
> vice versa, I would reach the limit at 30/30, NOT 15/15.
If you had 30 licenses, yes the limit would be when you needed either
30 decoders or 30 encoders. i.e. 1/30 would max you out.
-M+
__
aw to create these types of diagrams so we thought we
would give away software that is fully intended for this type of work :)
Thanks,
Matt G
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNS
- http://www.ioccc.org/
>
Interesting. I hadn't heard of this before. I know you're somewhat joking,
but there's some truth to this as well - I've seen some pretty obfuscated
dialplans in my day. I'll keep it in mind, h
esitate to contact us to address them.
Now go get designing! We wish you all the best of luck and look forward to
seeing all of the great designs!
Submit your phone system IVR diagram entries either at the Rate My Dial Plan
website (http://www.ratemydialplan.com) or by e-mail at
[EM
ounds louder than the inbound call volume) as to make
> things unusable.
>
> Any ideas? At all? I'm still relatively new to the
> Asterisk-interconnected-to-PSTN side of things, and it seems like there are
> dozens of config files and tools so explicit instructions are
eed:
http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windows-mobile-6x-for-free-voip-calls-using-asterisk/
OCG;
Have you managed to get this working on the front speaker? Or still the back
speaker only?
Thanks,
Matt G
: http://www.voipphreak.ca
: http://ww
're looking for.
Thanks,
Matt G
: <http://www.voipphreak.ca> http://www.voipphreak.ca
: <http://www.ratemydialplan.com> http://www.ratemydialplan.com
: <http://www.asterisk-jobs.com> http://www.asterisk-jobs.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
We're using it here on dynamic IP from our ISP.
They provide reverse DNS, which we've simply setup a CNAME to.
So, CPE390480Q239432098423.MYISP.COM is cnamed to PBX.MYBUSINESSDOMAIN.COM
Did not have to change anything else for this to work.
Thanks,
Matt G
: http://www.voipphreak
On 11/21/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
> > On Fri, Nov 21, 2008 at 10:42 AM, Matt Florell <[EMAIL PROTECTED]> wrote:
>
>
> >> just keep in mind that in
> >> my opinion the 1.4 tree did not become usable until 1.4.18 when most
> >&
On 11/21/08, Tilghman Lesher <[EMAIL PROTECTED]> wrote:
> On Friday 21 November 2008 09:42:12 Matt Florell wrote:
> > On 11/20/08, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
> > > You also have
> > > people like Matt Florell who have continued to
ECTED]/tzafrir
> >
>
>
> I still compile and install 1.2 for the most part, for call centers
> and large systems.
>
> The few 1.4 installs that I have done have been for "medium" sized
> PBXs, say 50-70 phones/users and they have been trouble free for the
> mo
solve it. If I upgraded zaptel would this be fixed?
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://
uld be multiple
levels of geographic redundancy to eliminate any problems with a server
outage.
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asteris
wn to doing a:
> #include
> for a file that is not present.
>
> There was no error message on the console about it that I saw.
> However I removed more and more lines until something loaded.
> after that I narrowed it down to that #includes.
In the latest versions of Asterisk,
alls.
>
> I do some operation in dialplan depending the time using
> GotoIfTime(21:00-09:00|sat|*|*?context1|96|1)
>
> Have Queue App any option to resolve this that I´m forgetting?
You could create two queues with the same members. I.E. myqueue and
myqueue-r
d it's just for testing the BT200 or
BT201 will suit you fine. If you want to test more features, the GXP2000 is
relatively cheap.
Have you looked at Aastra? They offer some quality phones in the same range
as the GrandStreams.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://w
o with the provider.
Are you maybe seeing this:
http://bugs.digium.com/view.php?id=13597
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News
On 12/11/2008 6:20 a.m., Tilghman Lesher wrote:
> On Tuesday 11 November 2008 05:17:30 Matt Riddell wrote:
>> On 11/11/2008 10:48 p.m., samuel wrote:
>>> So far I've updated a few machines (1.4.22) and the DNS queries are
>>> reduced to a minimum, at least
I don't want to upgrade to 1.4.22?
Olle? Maybe? :)
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/
ver to 127.0.0.1 in resolv.conf
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/
unknown water, how do you know the
> upgrade
> doesn't introduce something new? If it's been running, why fix what ain't
> broke?
Heh, try subscribing to the bugtraq mailing list. Most days have
security vulnerabilities in most systems. If your box is 100%
in
69635904176186202%
To be slightly more accurate :)
According to google's figures (July 2007 est.)
USA: 301,139,947
World: 6,602,224,175
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end soluti
th of a context, extension.
Use the Local channel:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/
On 6/11/2008 2:39 p.m., Pedram M wrote:
> Matt,
>
> Yep, I forgot to post that here is the extensions.conf:
>
>
> [10]
> switch => Realtime
According to
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions
this should be:
switch => Realtime/[EMAIL PROTEC
|
> | 10 | start | 3| WaitforSilence |
> +-+---+--++
>
>
>
> Any ideas on where to begin w/ the debug would be very appreciated.
Are you doing the switch from the dialplan in the [10] context?
Have a look at the realtime page on
ical ringers.
Isn't there a boostringer option for modprobe?
Not sure if that's what you're looking for though.
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.vent
urvey, I have no idea what the actual intent of the
> question was.
>
> Mark Michelson
Yeah, I just assumed that was what the question meant.
We're located in New Zealand but install systems worldwide (USA
included). Heh probably don't need the bra
what is happening with the call.
If you use the 'r' option then it will play ringing tones even if the
phone is busy.
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
Try using SSMTP
http://www.linux.com/articles/132006
It works with any provider for mail sending, and takes 30 seconds to setup.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
> -Original Message-
> From: [EMAIL PRO
On 5/11/2008 5:57 a.m., andrea wrote:
> Dear List
> I'm asking if there is a small hardware already implemented with
> software that just do traffic Shaping QoS?
> I have found *D-Link DI-102 VoIP QOS Adapter Packet Prioritizer
> *But it does not look available in Italy !
> Regards Andrea
The Link
f the Trixbox-generated dialplan
and utilities.
MATT---
On 10/29/08, Ron Byer Jr. <[EMAIL PROTECTED]> wrote:
> I noticed that the vicidial site has documentation available which probably
> covers the topics required. However, I also see that they want $50-$100 to
> download th
. Ah, I'm not positive on what would work for this - sounds like some
modifications to FOP may be in need. Maybe someone else on the list has
ideas.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
From: [EMAIL PRO
http://www.trixbox.org/forums/trixbox-forums/open-discussion/asterisk-guru-queuestats-install-guide-video
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark
mation you receive from
FOP.
Thanks,
Matt
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> Are you sending SMS to known users or to any mobile phone user?
>
> If you are sending to a fixed user base, track down the email to SMS
> gateways for their carriers. Then sending an SMS is no different than
> sending an e-mail.
>
If it's for something really important this might not
sterisk+sip+md5secret
>
> We use both secret= and md5secret= with the same password in each, one
> encrypted and one not encrypted - this seemed to let our 7970 register.
>
>
Matt,
I looked at this and did what it says for 1 of my phones.
Still no go.
I have a mix of polycom an
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, October 07, 2008 5:42 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] cisco phones getting SIP 401 unauthorized
Matt,
The phones are inside the LAN.
what is
traversal over a
firewall - are the phones inside the lan or on the net?
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, Oct
This may be what you're looking for:
http://www.linuxjournal.com/content/custom-checks-and-notifications-nagios
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROT
Ah, I don't use the touchflow crap :)
On mine on the today screen (you'll have to go to settings, today, items)
Set "internet telephony" to "on" and you should see it on the home screen.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
I use the TytnII with Win Mob 6.1, customized ROM and it's working for me -
through the back speaker though.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROT
can't
remember the name now) that works through the back speaker, but it's
payware, and nowhere near as lightweight as this method. For now I deal with
the back speaker.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
-Original Me
Hello,
If you are looking for a list of Call Center software packages that
work with Asterisk then take a look here:
http://www.voip-info.org/wiki/view/Predictive+dialer
There are over 20 now I believe.
MATT---
On 10/1/08, broadband Voice <[EMAIL PROTECTED]> wrote:
> I stumbled
functionality as
compared to the full agent interface.
Thanks,
MATT---
On 10/1/08, broadband Voice <[EMAIL PROTECTED]> wrote:
> Can I used Aastra phones as agents instead of web-base on
> astGUIclient-VICIDIAL suite: 2.0.4? Thanks. Our Asterisk is remote and call
> center will be usin
Check our howto:
http://www.voipphreak.ca/2007/04/16/monitoring-asterisk-14-with-snmp-and-cac
ti-for-pretty-graphs/
and for nagios monitoring
http://www.voipphreak.ca/2008/06/19/monitoring-asterisk-with-snmp-nagios-and
-nagios-administrator-using-ubuntu-lts-804-server/
Thanks,
Matt G
: http
Do you have ztdummy loaded in the VM?
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael J.
Liberatore
Sent: Wednesday, September 24, 2008 8:28 PM
To: Asterisk Users Mailing List - Non
uding retransmissions
and timers for these, is well documented in the IETF
RFC 3261.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd Reese
---
Internal Access:
*98 for Personal Voicemail or *99 for Main Voicemail
You could easily append this to the subject line, so it will show different
per category.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
a bit of queue_log QueueMetrics integration in the
VICIDIAL project, and there are several things that QM can do now that
are beyond what basic queue_log entries out of base Asterisk can
provide, such as adding call status codes and agent pause codes, and
several other features.
MATT---
_
-Original Message-
From: John covici [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 11, 2008 1:52 PM
To: Matt Gibson
Cc: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [asterisk-users] dahdi vs zap with latest version of asterisk
ztd-loc.ko -> dahdi_dynamic_loc.ko
ztdummy.ko -> dahdi_dummy.ko
ztdynamic.ko -> dahdi_dynamic.ko
zttranscode.ko -> dahdi_transcode.ko
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.c
ot;dhadi show channels"
*CLI> dahdi show channels
Chan Extension Context Language MOH InterpretBlocked
State
As I said we have no analogue connectivity to test with, so it could be
working, or could not be for us :/ I'll see if I can dig up a phone to test
with.
Tha
ned it earlier, but it may have gotten lost on the thread. I
was having problems with Dahdi until I added echocancel to our system.conf,
could this be your problem?
As soon as we added the echocanceller we were good to go.
echocanceller=mg2,1-4
Thanks,
Matt G
: http://www.voipphreak.ca
: http:
What do you get when you type show features?
On 9/6/08, Mark Hamilton <[EMAIL PROTECTED]> wrote:
> Hi James,
>
> Thank you very much for a detailed reply. (Matt, sorry about earlier, I
> totally missed the part you said about the t option)
> To answer, yes the Queue comm
t;
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Sent from Gmail for mobile | mobile.google.com
Matt Riddell
Director
VentureVoIP
___
-- Bandwidth an
>
> I have the same question as:
>
> http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html
>
> ..which like all important things was never answered.
>
>
>
> How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it'
Let me know if you find out - We played around with this for a while but
could never get it to work. We ended up sending multiple messages with blank
lines to get the spacing we wanted.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
-Original Message-
From
Hi JR,
This may help you - we were using it to route calls from friends through the
IVR so they hit us directly. You'll have to modify it to suit your dialplan,
but it should be a good starting point.
http://www.voipphreak.ca/2006/11/26/asterisk-14-php-rolodex-howto-script/
Thanks,
M
EUS Networks
> Office: 212-624-5943
> Web: www.euscorp.com
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Regi
I just wanted to note that we do have a very active community forum
for VICIDIAL available at:
http://www.eflo.net/VICIDIALforum/index.php
MATT---
On 8/20/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
> You need to install the MySQL client libraries and MySQL driver for
&g
I;m using Aastra 480i's 9133i's, 9112i's, and 57i's and none of them have
experienced problems with qualify=yes.
I;m currently on Asterisk 1.4.17, but I've also tested them with 1.4.14 up
to 1.4.19.
--
Matt
http://www.mattgwatson.ca
On Fri, Aug 15, 2008 at 10:5
ether) I can't see
how you would get that to work even if it is possible in TBCT.
I believe that only DMS100, NI2 and 5ESS PRI signalling protocals are
capable of TBCT with the current zaptel code-base. Also, the two B
channels involved i
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Darrick Hartman wrote:
> Matt Riddell wrote:
>> -BEGIN PGP SIGNED MESSAGE-
>> Hash: SHA1
>>
>> Paul Hales wrote:
>>> That's a good question - the plantronics are available with
>>> interchang
if
you use the plantronics (plugged into the handset thing) you still need
to take the phone off the hook.
Unlike the snom 360 where there is a separate socket.
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end t
ibank
> 8 fxs. It is running Zaptel 1.4.11 and Asterisk 1.4.18.
Hi,
I had a similar problem a while back with a 2N box. For some reason
there was a very loud click which caused the echo canceller to lose the
plot.
Have you tried disabling the echo can on that port?
- --
Kind Regards,
ission succeeded
FAILURE Transmission failed
UNSUPPORTED Text transmission not supported by channel
At this moment, text is supposed to be 7 bit ASCII in most channels.
The option string many contain the following character:
'j' -- jump to n+101 priority if the channe
ther presentations I'm giving there, if you
happen to be going.
MATT---
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk
ther presentations I'm giving there, if you
happen to be going.
MATT---
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk
If that's the only set of errors it might be a PHP/Apache error.
I would recommend posting on the VICIDIAL Forums if you can't get it to work:
http://www.eflo.net/VICIDIALforum
MATT---
On 8/8/08, Brad <[EMAIL PROTECTED]> wrote:
> Warning: Cannot modify header informatio
We have done this several times for customers with VICIDIAL. I have
also seen companies use AGI scripts to enable this kind of application
as well. So, yes it is possible.
MATT---
On 8/7/08, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> CRM: Customer Record Module which is any kind of ap
how I can fix this?
Thanks and regards,
Patrick
Did you change your SEPXXX when you upgraded to 8.3.3? You may have to
revert those changes. Check the debug log on the phones web interface to see
if it's choking on a particular line in the cfg.
Thanks,
Matt G
: http://www.voipphreak.c
that has left the company. I do have telephony
> experience with Legacy systems.
>
> Any help is appreciated.
>
As a temporary work around while you resolve the larger issue, you could
install something like mailx or ssmtp to relay over any smtp serv
I've used http://www.pbxprompts.com/
The whole pack is around 100$ and then I think I was charged 11$ per prompt
for custom ones. No setup fee that I recall.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
From: [EMAIL PROTECTED]
[mailto:[
Hi All,
Apologies for this, migrated the site and forgot to change a path. Site's
back up now.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Sunday, Ju
I've seen it before infact there is a website setup where people can
post stuff made with it... kind of super nerdy!
http://www.ratemydialplan.com
--
Matt
http://www.mattgwatson.ca
On 7/27/08, Peter Lindquist <[EMAIL PROTECTED]> wrote:
>
> Dean Collins wrote:
>
>
Call me crazy, but why are you so keen on selling them an Asterisk box
when you don't even know if its capable of doing what you want to sell
it for?
thats kinda scray actually.
--
Matt
http://www.mattgwatson.ca
On 7/22/08, voip crazy <[EMAIL PROTECTED]> wrote:
> Hello all,
>
o I can help ya there.
--
Matt
http://www.mattgwatson.ca
On 7/21/08, Noah Miller <[EMAIL PROTECTED]> wrote:
> Hi Joseph -
>
>> I have Astra 480i's and Snom M3's. I am using a SIP provider so I do
>> not have any peripheral cards.
>>
>> I am on
c:5839 set_destination:
> Can't find address for host '"72.16.1.20'
>
Might want to post a sip debug of one of the sessions from the Mitel phone.
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation Provided b
On 7/17/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
> Matt Florell wrote:
>
> > But seriously, Asterisk is a better example of doing things right more
> > recently, a couple of years ago all sorts of stuff went into the
> > "stable" releases of Aster
/06/fun-with-hpec/
--
Matt
On Thu, Jul 17, 2008 at 10:00 AM, Noah Miller <[EMAIL PROTECTED]>
wrote:
> Hi Loic -
>
> > According to that its using MG2.
>
> I think it will say MG2 regardless of whether or not there is a
> hardware module present.
>
>
> &
On 7/17/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
> Matt Florell wrote:
>
> > No apologies necessary, I think a lot of what you said is mostly true.
>
>
> Well, thank you. I really appreciate that you're willing to entertain
> what I am saying without c
12:52:55 nelson Completed startup!
Thats on my TE220B. the VPN450 is the hw echo can daughter board.
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 P
On Wed, Jul 16, 2008 at 3:07 AM, Gordon Henderson
<[EMAIL PROTECTED]> wrote:
> On Tue, 15 Jul 2008, Matt Darnell wrote:
>
>> Does anyone know of a bandwidth test that tests the upload with the download?
>>
>> All of the ones I can find will test the upload t
dering it, or at least earning the ire and distaste
> of many other list members, most certainly including the authors.
>
> But, that's no reason for self-censorship. So, with apologies to Matt
> Florell and others:
>
> Personally, I've found that ViciDIAL general
Hello,
You can try posting on the VICIDIAL forums asking for feedback from
average users:
http://www.eflo.net/VICIDIALforum/index.php
MATT---
On 7/16/08, Ein Bielaczyc <[EMAIL PROTECTED]> wrote:
> On Thu, Jul 17, 2008 at 1:35 AM, Matt Florell <[EMAIL PROTECTED]> wrote:
> &g
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Matt Darnell wrote:
> Does anyone know of a bandwidth test that tests the upload with the download?
apt-get install iperf
man iperf
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.
of VICIDIAL for companies from 1 to 300+ seats.
VICIDIAL is GPL and free, and there is a very active community support
forum. There are also paid support plans available.
Hope that helps,
MATT---
On 7/16/08, Ein Bielaczyc <[EMAIL PROTECTED]> wrote:
> I have a small customer looking
and downloads or
P2P sharing.
I would like something more formal that would keep the upload speed
the same as the download. VoIP as you know is symmetric.
The one VoIP test I find doesn't tell you how many calls you can
handle, just if it is VoIP ready.
und since before 2000.
I thought l0pht or someone did an article about it back around 2000.
Then they went and spoke at Whitehouse dinners and stuff and kinda
disappeared.
In those days I was heavily into greyhat and IDS systems, but I'm pretty
sure it was "common knowledge".
- --
K
of Asterisk, Jim Dixon, Mark Spencer, Steve
> Underwood, Nicolas Gudino, and I will leave off the fifth as to not leave
> anybody out ;)
Me, me, me!
:D
Or, Kevin, Russell, Olle, Josh, Critch (although he's been pretty quiet
lately), I guess the list goes on.
- --
Kind Regards,
M
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jason Dixon wrote:
> On Tue, Jul 08, 2008 at 11:00:43AM -0400, Jason Dixon wrote:
>> On Tue, Jul 08, 2008 at 12:10:05PM +1200, Matt Riddell wrote:
>>> Action: Command
>>> Command: show queue my_queue_name
>>> Action
ation, i am reading, the bug report to see,how i
>> can verify that i have the gsm bug.
>
> Well, if you have gcc version 4.2.x (you can check with "gcc -v")
> there's a good chance this is the problem.
Just do:
export CC=gcc-4.1
export CXX=gcc-4.1
./configure
make
Wo
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Loic Didelot wrote:
> Hello,
> I just got my Xorcom BRI bank. Seems to work. But I have some questions.
> Is anyone getting good values using zttest?
Is it plugged into the BRI?
Is it the sync master?
i.e. xpp_sync
- --
Kind Regards,
Mat
all control.
It checks the DB every few seconds, and updates the credit based on how
long the person has been talking and the rate to that destination.
They can add more money at any time, and the DB value is updated.
When they run out of credit the call is killed via the manager.
- --
Kin
701 - 800 of 4003 matches
Mail list logo
