minute walk
west on College or take the streetcar two stops to St. George.
There is a campus map available on the web at
http://www.osm.utoronto.ca/map.
If you require additional information, please contact Simon Ditner
[EMAIL PROTECTED], or myself, Jim Van Meggelen[EMAIL PROTECTED]
This event
is a similar phone with more line appearances, a higher
resolution display, full handsfree (Polycom-quality) and PoE. The IP500
has been favorably compared to the Cisco 7940.
Cheers,
--
Jim Van Meggelen
[EMAIL PROTECTED]
--
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. Kinda like a team of drunken Canadian hockey
players; friendly, but you'd never know it cause they're so dangerous.
Welcome to Asterisk-Users
--
Jim Van Meggelen
[EMAIL PROTECTED]
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]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jim Van Meggelen
Sent: Jueves, 10 de Marzo de 2005 12:17 a.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Comparison Charts
[EMAIL PROTECTED] wrote:
Guys.
Anybody has a URL or some document
have to
adopt a new way of thinking about telecom systems.
Welcome to Asterisk!
--
Jim Van Meggelen
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to include every US and CDN
NPA.
Try nanpa.com
This might help:
http://www.nanpa.com/nas/public/npasInServiceByLocationReport.do?method=
displayNpasInServiceByLocationReport
--
Jim Van Meggelen
[EMAIL PROTECTED]
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Version
techies have gone home
for the evening and I was looking for an answer before I left.
You could do that with two tin cans and a string! ;-P
In all seriousness, the answer to your question is: yes, Asterisk can
do that, and a whole lot more.
Cheers,
--
Jim Van Meggelen
[EMAIL PROTECTED
Thanks as always to everyone who provided feedback. It was most helpful!
Regards,
Jim.
--
Jim Van Meggelen
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I would like to start a discussion centred around the various
ways one might serve up configuration files from an Asterisk
server (I know
deprecated, and it does not appear to be actively maintained.
Is TuX the way to go?
As for tftpd and ftpd, I'm just not sure. Leightweight is the key, here.
Thoughts, opinions, experiences?
Thanks,
Jim.
--
Jim Van Meggelen
[EMAIL PROTECTED]
--
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Checked
designed for
voicemail integration (although you still won't get the CLID).
Jim.
--
Jim Van Meggelen
[EMAIL PROTECTED]
--
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[EMAIL PROTECTED] wrote:
I didn't realize that the stable branch was never added to...
So it will NEVER have any more features than it currently has???
1.0 STABLE will never have any more features.
1.2 STABLE will be released in the next 3 to 6 months, and it will
include all features that
[EMAIL PROTECTED] wrote:
I've added an introduction article about the ARA on my web
site http://www.voip-forum.com/
The same text is now also added to CVS head as
README.realtime. On the same site, you will also find the
news item about how we used
Asterisk for a call from an airline jet
.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Jim
Van Meggelen Gesendet: Freitag, 25. Februar 2005 05:57
An: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Betreff: RE: [Asterisk-Users] Transfer a call ? Am I looking
[EMAIL PROTECTED] wrote:
On Fri, 2005-02-25 at 00:50 +0100, Mateo Meier wrote:
Hey Guys
Im trying to forward a call with asterisk to a regular phone.
Something like I get a call on my regular phone, and he's trying to
reach some buddy of mine.. then I tell him wait a sec and push
Flash
easier, and having no moving parts makes them
easier to ruggetize.
This is just a brainstorm, so all thoughts are welcome.
TIA,
Jim.
--
Jim Van Meggelen
[EMAIL PROTECTED]
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, Jim Van Meggelen wrote:
Folks,
In light of all the troubles people report when running more than
one TDM400 card in a system, I wouldn't mind hearing what your
solution of choice would be when having to connect 5 or more analog
telco circuits to an Asterisk.
I'll try and compile
[EMAIL PROTECTED] wrote:
On February 20, 2005 09:25 am, Michael Giagnocavo wrote:
Sorry, I understood the O.P. already had the hardware bought and
installed and simply wanted to throw on an extra line.
You understood correctly;
Uh, nope. I've been unclear. It was a purely hypothetical
[EMAIL PROTECTED] wrote:
On 20 Feb 2005, at 16:22, Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
If you have a TDM card already, buying a T1, channelbank, etc. to
add a few lines is the stupidest thing I've heard of today.
Not necessarily stupid, but certainly expensive.
Have you
for their feedback.
Regards,
Jim.
--
Jim Van Meggelen
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Folks,
In light of all the troubles people report when running more
than one TDM400 card in a system, I wouldn't mind hearing
what your solution of choice would be when having to connect
5
[EMAIL PROTECTED] wrote:
Jim Van Meggelen wrote:
Yep, that's a possibility, but it's rather more kludgy than I'd like.
(heck, the channel bank and T1 is more kludgy than I'd like - an
integrated card would be my preference).
I haven't followed this thread closely but have you looked
[EMAIL PROTECTED] wrote:
On Sun, 2005-02-20 at 11:47 -0500, Jim Van Meggelen wrote:
Well, I appreciate everyone's input, and I'll give the matter some
more thought.
Just so no one stays up at night worrying, this is not a situation I
am facing, it is simply a hypothetical scenario
Title: Message
Just plug
it in. The RJ11 is narrower than the RJ48, but has the exact same connection
mechanism. it'll fit perfectly (the centre two pins are the
contacts)
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL
Van Meggelen
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Kohlsmith
Sent: Sunday, February 20, 2005 1:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [Asterisk-Users] Segmentation fault
[EMAIL PROTECTED] wrote:
Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
Jim Van Meggelen wrote:
Yep, that's a possibility, but it's rather more kludgy than I'd
like. (heck, the channel bank and T1 is more kludgy than I'd like
- an integrated card would be my preference).
I haven't
solution that I know of.
Jon.
On Friday 18 February 2005 09:21 am, Jim Van Meggelen wrote:
Folks,
In light of all the troubles people report when running more than one
TDM400 card in a system, I wouldn't mind hearing what your solution
of choice would be when having to connect 5 or more
, but I can't
find anything in the list archives or Wiki with clues as to how this
might be achieved.
Any advice would be appreciated.
Cheers,
Jim.
--
Jim Van Meggelen
[EMAIL PROTECTED]
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Version: 7.0.300 / Virus Database
into the Wiki, as
I figure this could be useful knowledge for the community.
TIA,
Jim.
--
Jim Van Meggelen
[EMAIL PROTECTED]
--
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Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 14/02/2005
You are using illegal characters in your file name.
See this line in your output?
ast_writefile: No such format 'wav|rec_to_448704386865_at_16022005-16'
It can't get past it because the colon is not a valid filename
character.
[EMAIL PROTECTED] wrote:
Hello,
I have been attempting to
OK, forget I said that. Wrong side of my brain.
Still, it is funny that it truncates the filename at the colon.
These lines are suspicious:
CLIast_writefile: No such format
'wav|rec_to_448704386865_at_16022005-16'
Where'd the :56:35 on the end go? Also, why is is trying to set the
format to
[EMAIL PROTECTED] wrote:
Hello,
There is no colon the filename below.
Exactly. But there *is* (or rather *was*) in the filename you told it
you wanted to write.
Where'd it go?
--- Jim Van Meggelen [EMAIL PROTECTED] wrote:
You are using illegal characters in your file name.
See
[EMAIL PROTECTED] wrote:
Hi!
Maybe I have just been looking on the wrong pages but there is a
question that is very important for me. I already studied some
Demo-Dialplans and made some basic experiences with Asterisk.
But what I
need to find out is how I can handle this.
I am leaving
[EMAIL PROTECTED] wrote:
Stefan Gofferje wrote:
Hi folks,
has anybody knowledge about the Nortel i2004? Nortel calls it
Internet Phone. I'm curious, which protocols it may understand...
I just came back from a Nortel roadshow and was told it's H.323.
That's one of the things that's so
[EMAIL PROTECTED] wrote:
Carl Sempla schrieb:
On Monday, 14 February, 2005 02:30 : George Cohn
[EMAIL PROTECTED]
wrote:
Stefan Gofferje wrote:
Hi folks,
has anybody knowledge about the Nortel i2004? Nortel calls it
Internet Phone. I'm curious, which protocols it may understand...
[EMAIL PROTECTED] wrote:
Nortel i2004 spec is available here.
http://www.nortelnetworks.com/products/library/collateral/i2004.pdf
Documentation implies H.323 and MGCP interoperability.
I'd love to know where to get the firmware for that.
I am not sure
of the nature of the relationship,
[EMAIL PROTECTED] wrote:
asterisk-users-bounces at lists.digium.com wrote:
Hi everyone,
I was working yesterday and after I provide my IAXy box it loose any
network comunication, the link light (green) is on and the activity
light (orange) when the power is turned on it does nothing, but
[EMAIL PROTECTED] wrote:
Hi everyone,
I was working yesterday and after I provide my IAXy box it
loose any network comunication, the link light (green) is on
and the activity light (orange) when the power is turned on
it does nothing, but when I pickup the phone connected to the
box, this
[EMAIL PROTECTED] wrote:
Hi,
I'm just wondering how other people cost/charge for * boxes
they have installed in small businesses?
Asterisk is such a complex beast and is capable of so much it
seems equally complex to figure out a charging model!?
I agree.
I'm contemplating installing a
.
Asterisk - /etc/asterisk/zapata.conf
Linux - /etc/zaptel.conf
Or, hierarchically:
[Asterisk]
(/etc/asterisk/zapata.conf)
^
|
v
(/etc/zaptel.conf)
[Linux]
Not well explained by the documentation, perhaps, but logical
nonetheless.
Cheers,
Jim.
--
Jim Van Meggelen
[EMAIL PROTECTED
[EMAIL PROTECTED] wrote:
Are there any VOIP lobbyist groups in Canada?
Well, he's not exactly *in* Canada, but there's Jeff Pulver:
http://www.crtc.gc.ca/ENG/transcripts/2004/tt0922.htm
He seems to be about all we've got, thus far.
-Original Message-
From: [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I am looking at setting up the following configuration and any
help/input/comments before signing the PRI contracts will be greatly
appreciated.
PRI Tampa PRI
Sarasota PRI
-- Nortel BCM--Asterisk--Asterisk---
[EMAIL PROTECTED] wrote:
Hi community,
does anyone out there made some experience with Varion
(www.govarion.com) based E1/T1 cards ?
I have no hands on experience with them, but they are based on the
open-source Tormenta card:
www.zapatatelephony.org
With Asterisk, I think any hardware
[EMAIL PROTECTED] wrote:
does anyone know when current HEAD is scheduled to stabilise? Is
there a plan, or is it still some time in the future?
I believe I saw an announcement recently that it will start
stabilizing in February, with the goal of releasing 1.1 on the
six-month anniversary of
Dave Morrow wrote:
Hi all, I have asterisk installed and working just fine with a couple
of Cisco IP Phones. I am now ready to pilot connectivity to PSTN and
am wondering what hardware would be recommended to make minimum
connectivity to the public telephone network.
Minimum would be any
data on Linux for years, and they're recently added zapata
to their list of open-source drivers.
Give them a shout, they love this kind of stuff.
Cheers,
Jim
--
Jim Van Meggelen
[EMAIL PROTECTED]
--
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Version: 7.0.300 / Virus
[EMAIL PROTECTED] wrote:
Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
[snip]
Your diagram is a bit confusing to me. Still, the use of PRI in the
BCM is a good plan. You've avoided using the highly unstable VoIP
functions in the BCM.
Sorry, it got a little jumbled
No worries. I
[EMAIL PROTECTED] wrote:
Hello,
I am new here. I am also somewhat new to telephony and IT,
however, I am technically adept.
That's probably the most important thing with Asterisk.
I work for a small but growing non-profit in Colorado and I
wear many hats. Because I am the only one who has a
[EMAIL PROTECTED] wrote:
Jim Van Meggelen wrote:
I had always thought that the difference between STABLE and HEAD
*was* the stability. Not so much stability in terms of code, but
more in terms of features and compatibility. Is that not the same
numbering philosophy that applies with Linux
[EMAIL PROTECTED] wrote:
Roy Sigurd Karlsbakk wrote:
does anyone know when current HEAD is scheduled to stabilise? Is
there a plan, or is it still some time in the future?
I believe I saw an announcement recently that it will start
stabilizing in February, with the goal of releasing 1.1
[EMAIL PROTECTED] wrote:
Jim Van Meggelen wrote:
So 1.0.x STABLE will become 1.2.0 STABLE, and 1.1.x HEAD will be
continued as 1.3.0 HEAD
Could be wrong, but it'd make sense.
It would make sense if we ever did unstable releases from HEAD, but we
don't do that currently :-)
I had
[EMAIL PROTECTED] wrote:
Hi,
I am considering A101/102/104 cards for my asterisk
installations. Has any of you used these or any Sangoma cards
in such environment? Any thoughts? How do they stack up
against Digium cards? Any input would be greatly appreciated. robert
The Sangoma cards are
Title: Message
www.voip-info.org
www.asteriskdocs.org
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vassili
GontcharovSent: January 24, 2005 5:13 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] PSTN and
Asterisk
knowledge of any organization that
might maintain such stats.
Regards,
Jim.
--
Jim Van Meggelen
[EMAIL PROTECTED]
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.
--
Jim Van Meggelen
[EMAIL PROTECTED]
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___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
Title: Message
I tested
this as well and found the same thing.
I
have submitted a bug report:
http://bugs.digium.com/bug_view_page.php?bug_id=0003394
That
one's severe enough that I have flagged it as a major. On a busy system, this
bug could cause all kinds of messages to go missing,
[EMAIL PROTECTED] wrote:
Hi All,
For a small installation using ITSPs via DSL is G.729 a
worthwhile exercise? I have G.729 capable SIP phones and my
ITSPs cupport the codec so I could go end-to-end without
transcoding. What's call quality like compared to G.711, GSM or iLBC?
Low bandwidth
[EMAIL PROTECTED] wrote:
Low bandwidth
Low CPU utilization
Best audio quality
I think you might want to clarify that Best audio quality is in
relation to other highly compressed codecs. Certainly my (albeit
limited) experience is that g711 is much more clear than g729.
Compared
[EMAIL PROTECTED] wrote:
According to CarrierAccess, the Adit 600 uses CAS for voice
signalling. What is this? This should not be a problem for
Asterisk? Does the Asterisk server need to reencode CAS into aLaw
when going to Euro ISDN?
Try this:
), something I wouldn't feel was appropriate on Asterisk-Users.
Time will tell how many topics there are to discuss.
The way I see it, a Canadian mailing list will be no different than our
country itself: visitors will always be welcome.
--
Jim Van Meggelen
[EMAIL PROTECTED]
--
No virus found
[EMAIL PROTECTED] wrote:
Hello All,
Please forgive the lack of understanding as of yet but I have
been trying to follow the mailing list messages over the last
few days and would like to know if someone could wither point
me into the right direction or possibly give me a brief
overview of
.
As VoIP becomes ubiquitous, the value of regional lists might be less
and less relevant.
Jim.
-Original Message-
From: Jim Van Meggelen [mailto:[EMAIL PROTECTED]
Sent: Monday, January 17, 2005 11:42 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk
[EMAIL PROTECTED] wrote:
On January 17, 2005 11:42 am, Jim Van Meggelen wrote:
Anyhow, I signed up, and am planning to start a thread about the CRTC
VoIP deliberations (and the generous act performed there by Jeff
Pulver), something I wouldn't feel was appropriate on Asterisk-Users.
Time
[EMAIL PROTECTED] wrote:
On January 17, 2005 01:57 pm, Jim Van Meggelen wrote:
I'd have a hard time appreciating regulatory challenges in regions
that I don't have involvement with, so I assume that the reciprocal
is generally true as well. Granted, I shouldn't presume that everyone
thinks
[EMAIL PROTECTED] wrote:
On January 17, 2005 04:47 pm, Jim Van Meggelen wrote:
LOL. I hadn't thought of it that way. Little vignettes amidst the
commercials?
Exactly -- It's precisely why I hang around on linux-elitists
and a couple
other oddball lists... a good 90% of what's
Kim Lux wrote:
I called Telus before Christmas requesting some sort of VOIP
connection. Here is what I learned:
a) the guy I was talking to never heard of *
That'll change.
b) they didn't think there was any way that a PC could
perform the duties of a PBX
He was probably thinking of the
[EMAIL PROTECTED] wrote:
Quoting [EMAIL PROTECTED]:
Would it be considered trolling to start a thread on Cleaning Maple
Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi?
Let's not forget the weekly tooques and telephony segment, and a
review of the best block heaters for your wi-fi
[EMAIL PROTECTED] wrote:
Since i need to run at least 30 FXS-channels, what hardware
should I use? Both Motherboard etc and FXS-hardware? There
seems to be only 4-channel cards from digium. Then I would
need at least 8 cards. 8 FXS-cards + 1 E1-card. What
motherboard handles this? And to they
Title: Message
Is there a
dialtone recording?
I'm
thinking :
exten =
s,1,background(dial-tone-recording)
exten =
s,2,goto(,s,1)
I have not
tested this, but I can't see why it wouldn't work. The only problem would be the
lack of a timeout (and possibly a small challenge getting a
[EMAIL PROTECTED] wrote:
Hi;
I'm trying to connect a TDM400P with an FXS module to a Valcom V-9940
Paging adaptor. This port on the TDM400P was connected to a 2500 Set
and was working I just re-connected it to the Valcom (which
is known to
work on a Telco POTS line) and its not picking up.
Peter Svensson wrote:
On Tue, 11 Jan 2005, Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
I'm trying to connect a TDM400P with an FXS module to a Valcom
V-9940 Paging adaptor. This port on the TDM400P was connected to a
2500 Set and was working I just re-connected it to the Valcom (which
Rich Adamson wrote:
I'm trying to connect a TDM400P with an FXS module to a Valcom
V-9940 Paging adaptor. This port on the TDM400P was connected to a
2500 Set and was working I just re-connected it to the Valcom (which
is known to work on a Telco POTS line) and its not picking up. The
Valcom
[EMAIL PROTECTED] wrote:
Hi,
I have Siemens combiset - it can gateway GSM phone to normal
analog phone. It has output where I can connect regulat analog phone.
That would be an FXS connection.
How can I connect to combiset with Asterisk - via fxo or fxs ?
You would need to connect the
Anyone in the Toronto area interested in getting together to share notes
and swap war stories?
--
Jim Van Meggelen
[EMAIL PROTECTED]
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Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 06/01/2005
-Commercial Discussion asterisk-users@lists.digium.com
Sent: Saturday, January 08, 2005 12:37 PM
Subject: Re: [Asterisk-Users] Toronto?
On Sat, 8 Jan 2005 05:40:02 -0500, Jim Van Meggelen
[EMAIL PROTECTED] wrote:
Anyone in the Toronto area interested in getting together to share
notes
César Davi Ávila do Nascimento wrote:
Hi All
I've bought a TDM400P and need some help with configuration. Can you
tell me what to do ?
I've tried to install and the message below has appeared:
[EMAIL PROTECTED] asterisk]# modprobe zaptel
[EMAIL PROTECTED] asterisk]# modprobe wcfxo
[EMAIL PROTECTED] wrote:
This has been an interesting discussion. I'll chime in with my
experience here.
I have two servers. One with the cheapest motherboard and athlon
processor I could find on Newegg.com. The other is a 1999 era
motherboard with a Via C3 processor, again a bargain
Dorn Hetzel wrote:
On Sat, Jan 01, 2005 at 07:23:58PM -0500, Jim Van Meggelen wrote:
[...]
What if, for example, the TDM400 issues were a cumulative thing? If
you had over 6dB of attenuation on the PSTN loop, coupled with
greater than 5V potential on the neutral-ground of your elecrical
-conditioned UPSs?
Hope that helps-
T
Jim Van Meggelen wrote:
Dorn Hetzel wrote:
On Sat, Jan 01, 2005 at 07:23:58PM -0500, Jim Van Meggelen wrote:
[...]
What if, for example, the TDM400 issues were a cumulative thing? If
you had over 6dB of attenuation on the PSTN loop, coupled
Title: Message
Please put
this in the Wiki (www.voip-info.org). It
is valuable information and will reach a wider audience
there.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: January 3, 2005 1:52
AMTo:
[EMAIL PROTECTED] wrote:
On Sat, 2005-01-01 at 16:14 -0700, Michael Welter wrote:
Since you asked, and since I'm well into this bottle of Merlot on
New Year's day:
1. Power alarms. WTF does that mean? Wish I had some support docs.
2. On bootup, Excessive leakage module x, ProSLIC
[EMAIL PROTECTED] wrote:
Paul Fielding wrote:
Hmmm I could certainly see that being the issue. If it is the
issue, though, then I think it's something that needs to be
addressed.
In my opinion, Digium needs to address it, as well as the whole
provisioning via cli thing. I know
You need two files to make the zapata stuff work:
/etc/zaptel.conf
/etc/asterisk/zapata.conf
The first one (/etc/zaptel.conf), configures the Linux driver for the
hardware. In theory, you could have another application other than
Asterisk using the zaptel driver.
The second one
[EMAIL PROTECTED] wrote:
Hey gang,
I'm looking at escaping from a Nortel Meridian CISC system
to Asterisk/Digium/SIP phones. I'm currently in the testing
and proof of concept phase. I'm going to need a SIP phone
and don't want to re-purchase and have orphans around.
I've got a few
[EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield Sent: Saturday, 4 December 2004 1:44 AM
To: Ed Greenberg; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Ouch, part
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield Sent: Saturday, 4 December 2004 1:44 AM
To: Ed Greenberg; Asterisk Users Mailing List - Non-Commercial
Discussion Subject: Re:
[EMAIL PROTECTED] wrote:
On December 21, 2004 08:29 am, Jim Van Meggelen wrote:
5 Watts per FXS
They also told me each FXS port would support a REN (ringer
equivalence number) of 5.0, which means that you should be able to
to ring five of the old electromechanical telephones simultaneously
[EMAIL PROTECTED] wrote:
On December 21, 2004 08:29 am, Jim Van Meggelen wrote:
5 Watts per FXS
They also told me each FXS port would support a REN (ringer
equivalence number) of 5.0, which means that you should be able to
to ring five of the old electromechanical telephones simultaneously
[EMAIL PROTECTED] wrote:
hi
any chance of making asterisk support these?
http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-3835624908
8.htm
According to the manufacturer, they already do:
http://www.ipvolution.com/
Cheers,
Jim.
--
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[EMAIL PROTECTED] wrote:
Rich Adamson schrieb:
In the past I had problems with the audio over sip. Then I tried the
-p Option and increased the memory. Now it is better but not
perfect.
Are there any more possibilities to increase it more? By now I'm
using a P-II/333.
Could a
[EMAIL PROTECTED] wrote:
Hi Noah, I have been contacted by 2 people but nothing so
far. If you want to add $500 please email me your details and
I'll add it to the wiki to co-ordinate this.
I agree I'm really surprised why no one has shown more of an
interest in video calls on asterisk yet.
[EMAIL PROTECTED] wrote:
Jim Van Meggelen schrieb:
[EMAIL PROTECTED] wrote:
- ADSL 1000kBit/s Downstream, 128kBit/s Upstream
That upstream bandwitch will need to be managed carefully.
I know.
If you're using G.711, one channel would be using roughly 80kbit of
your upstream. Who has
[EMAIL PROTECTED] wrote:
Thank you all for your explanations related to my question.
I have one follow-up question though.
When I said that all dsp related stuff has to be handled
by software within asterisk, I was thinking of
conferencing at the time.
I mean in order to be able to
[EMAIL PROTECTED] wrote:
Gents,
Just a passing thought... is there any reason why the ability
to search the past posts on here isn't switched on?
Just wondered, since it makes much more sense to be able to
search the old archives if you have a problem, rather than
ask the same question
[EMAIL PROTECTED] wrote:
I am looking to help out my company find a more budget
conscious but reliable way to hold conference calls between
5+ people. 4x a month we hold several hour long conference
calls during non-business hours. All of the employees have
high speed internet. Currently
[EMAIL PROTECTED] wrote:
Sorry for the misspelling... Thanks for the replies. I will
set it up and start playing. This is all very exciting.
I've been using VoIP as my primary phone but this is going a
bit further. At the office we have a T1 that is probably
fairly dead after hours.
[EMAIL PROTECTED] wrote:
--On Thursday, December 16, 2004 3:59 PM -0500 Jim Van Meggelen
[EMAIL PROTECTED] wrote:
I've always found Newton's Telecom Dictionary to be a great
reference. It's not too technical, packed with humour, and very
comprehensive.
I have a very old copy
[EMAIL PROTECTED] wrote:
Antony Stone schrieb:
On Wednesday 15 December 2004 21:26, Michael Vogel wrote:
Is it a little bit too much for such a machine? What could be the
bottleneck? CPU? Memory? Interrupts?
My advice would be to whack in a load more RAM - basically, try to
get the poor
[EMAIL PROTECTED] wrote:
Jim Van Meggelen schrieb:
Hmmm. I propose that we make your system the list guinea pig! If we
can get that one tweaked, there's no telling what alse we can do with
Asterisk!
I'm not really sure that I completely get the meaning of this
sentence. (Which can
[EMAIL PROTECTED] wrote:
Jim Van Meggelen schrieb:
You understand. I was kidding a little bit, but yes, I am also
wondering just what things can be done to get a slower machine to
work as well as possible.
Okay. So lets try.
What are you running in terms of a kernel or distro?
Have you
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
I was posed this question:
A T1 set up for voice carries 24 conversations on a circuit that is
1.544 megabits/second. Right?
Yes and no. If the T1 is channelized, then yes. If it's a PRI
circuit, then it has
[EMAIL PROTECTED] wrote:
I was posed this question:
A T1 set up for voice carries 24 conversations on a circuit that is
1.544 megabits/second. Right?
Yes and no. If the T1 is channelized, then yes. If it's a PRI circuit,
then it has only 23 channels to carry voice, as the 24th channel is
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