same wonder here, cos I wasn't able to compile zapata on a mppe patched
kernel which includes ppp, but a patched version for pptp. I had tu
uncomment this line.
- Original Message -
From: "Adam Goryachev" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, April 04, 2003 4:57 AM
Subj
> first off the majority of work on the cards was made by zapata guys, let's
> give them the credit. secondly are you talking about hardward support or
> asterisk software support?
Wrong. The Tormenta2 was truly a joint venture between us. Digium
*totally* funded its development, *and* I persona
Just use the "Ring" application, typically followed by "Wait"
Mark
On Thu, 3 Apr 2003, Petr Michálek wrote:
> Michael Bielicki wrote:
>
> >Is it possible to give a false ringbakc on asterisk ?
> >
> >
> What hardware are you using?
>
> Petr Michalek
>
> __
We don't have stickers for the part 68 on the T100P yet because we haven't
done part 15 or UL on it yet.
Mark
On 3 Apr 2003, Steven Critchfield wrote:
> On Thu, 2003-04-03 at 10:55, Mark Spencer wrote:
> > > 1. it was FCC, CA and CE certified (FCC and CA states no card is reg with
> > > them as
I don't know about the database, but remember it may be listed under
"Linux Support Services" since we were not named Digium at the time.
Mark
On Thu, 3 Apr 2003, d hinton wrote:
> sorry if you took it wrong, i was just repeating what FCC told me. not
> trying to call you or your team liars. if
I am interested let me know whats the quantity you have.
Hemant
- Original Message -
From: "Benjamin J. Bawkon" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, April 04, 2003 3:51 AM
Subject: [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #239 - 13 msgs
> If anyone is int
d hinton wrote:
FIRST LET ME STATE AGAIN I'M NOT A TROLL! my small company would like to
support the asterisk effort, but can't or won't pay a $800 markup. now,
please go to the http://zapatatelephony.org/ website and do some reading!
the pci card was their last card released, BY THEM not digium,
No response on this post from a couple of weeks ago,
so I thought I'd send it out again. Any thoughts on
DTMF on inbound Iconnecthere calls? Does anyone have
this working?
Thanks again.
--- DA <[EMAIL PROTECTED]> wrote:
> It would be awesome to have incoming DTMF work with
> Iconnecthere. Has
I posted a message a little while ago but got no response (that I can
recall), I've also seen other people mention this issue.
Basically, when you have music on hold, it doesn't play the music on hold,
the debug info shows it is starting and then stops straight away..
# My extensions.conf ...
ex
Just wondering if there is a reason PPP support is compiled into zapata by
*default*:
# Uncomment for Generic PPP support (i.e. ZapRAS)
#
KFLAGS+=-DCONFIG_ZAPATA_PPP
Especially since the comments imply that it should be commented out by
default...
The main reason I ask is because I usually try to
Yes it is. Download it and look at the LICENSE file. It is the GPL. As for the zapatel
cards, all materials needed to manufacture the cards are online. They too are under
GPL.
"The better you look the more you will see." - Ms. Lidia
Gene
-Original Message-
From: d hinton [mailto:[EMAIL
FIRST LET ME STATE AGAIN I'M NOT A TROLL! my small company would like to
support the asterisk effort, but can't or won't pay a $800 markup. now,
please go to the http://zapatatelephony.org/ website and do some reading!
the pci card was their last card released, BY THEM not digium, inc. these
cards
> possibly a little more priority when asking questions to Mark in an
> online forum. His help has been invaluable for diagnosing problems and
> pointing towards better solutions. Then you can add in the fact that
> Mark can continue for that mutch longer doing asterisk development as
> his job.
T
Anyone know how many RENs an S100U can drive? Or where I can get
some sort of booster? I'm trying to drive 2 gong-type ringers (IE,
2.0A REN).
- awh
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On Thu, 2003-04-03 at 15:32, d hinton wrote:
> first off the majority of work on the cards was made by zapata guys, let's
> give them the credit. secondly are you talking about hardward support or
> asterisk software support?
I may be a little off, but I'm pretty sure the original was a ISA card
a
If anyone is interested in X100P Cards, I have quite a few brand new
ones. Never been opened. $55.00 each, and I'll include ground shipping
anywhere in the U.S.
If you want one or more, email me off list at [EMAIL PROTECTED]
And since the discussion has been raging, if you are looking for
suppo
Hi All...
I am having an odd problem. I had installed Asterisk on a test box several
weeks ago. Yesterday we moved it to a production box. Everything works,
except music on hold. I have verified that we are using mpg123 and not
mpg321.
The music is heavily distorted. It sounds like white
Playback does not take an extension of the file. It looks for the best
file format to use. Try using it without the .mp3 extension on there.
James
On Thu, 3 Apr 2003, Tamas Levente wrote:
> This is why I asked. (the file is there)
>
> DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Set
If you use playback you don't specify the file ending. Since I only use
the mp3 player for streaming content, I'm not sure if it holds true
there also.
On Thu, 2003-04-03 at 15:01, Tamas Levente wrote:
> This is why I asked. (the file is there)
>
> DEBUG[5126]: File chan_sip.c, Line 2773 (check_u
You could configure the channels of port 1 as
clear=1-24
in zaptel.conf
This way you'll have one big pipe on /dev/zap/1 accessible for you.
Unfortunatelly you cannot do clear=1-96 and have it all 4 spans
on /dev/zap/1.
regards
Martin
On Thu, 3 Apr 2003, Chris Albertson wrote:
>
> Is there a blo
first off the majority of work on the cards was made by zapata guys, let's
give them the credit. secondly are you talking about hardward support or
asterisk software support?
hi
let's deal with drivers first: the original was released by zapata guy's as
GLP'ed software and any derivertive of that w
This is why I asked. (the file is there)
DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP to
0
-- Executing Answer("SIP/levisnom-c18d", "") in new stack
-- Executing Playback("SIP/levisnom-c18d", "/asterisk/c.mp3") in new
stack
WARNING[16400]: File file.c, Line 410
digium cards on * but the carrier connections are all h323 and some of them
don't provide ringback.
On Thursday 03 Apr 2003 21:42, Petr Michálek shaped the electrons to say:
> Michael Bielicki wrote:
> >Is it possible to give a false ringbakc on asterisk ?
>
> What hardware are you using?
>
> Pet
James,
> I have to disagree here. I send and receive faxes over IAX all the time
What echo_cans are you using on each end and do you have daggressive
suppression enabled (one or both ends)?
John
This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus.
___
I've personally had nothing but problems getting FAX to work over IAX.
Some of my problems were resovled by tweaking echo cancellation and
jitterbuffer (I turned them both off essentially). Keep in mind this is
on a local 100Mbit ethernet switch. I can only imagine performance
degrades as latenc
I too.
We have a telco provider that brings analog
lines by SIP (converted to analog by
a cisco router in our office)
And we have a standard fax running on that
line
so...
remote FAX - GW -..sip..- CiscoGW - our fax
Matteo.
On Thu, 2003-04-03 at 21:28, James Golovich wrote:
> I have to disa
>disable echo cancellation
Can we currently do that on a per-channel basis? For our situation. * to iax
to * to pbx, I need it on - as most of the traffic is voice.
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian Jones
Sent: Thursday, April 03, 2
Michael Bielicki wrote:
Is it possible to give a false ringbakc on asterisk ?
What hardware are you using?
Petr Michalek
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Hello,
I have installed an asterisk server with 7960 sip phones and an eicon card
(diva 4bri server, I use only three channel) using capi channel.
Everything seems to work fine, but from time to time (two or twice a day)
during a call I get a capi error in asterisk log and asteris
Nice phone, but seems it's not supported in US anymore.
Haven't tested it with *.
alejandro
Clay Graner said:
> Here is a SIP phone I haven't seen before. Does anyone have any
> experience with this one?
>
> -Original Message-
> From: George Richardson [mailto:[EMAIL PROTECTED]
> Sent: W
On Thu, 2003-04-03 at 12:03, Jeremy McNamara wrote:
> Karl Putland wrote:
>
> >Well there's T.30 which is a store and forward mechanism, or T.38 which
> >is a realtime fax relay but they cost $ for licensing of the protocols.
> >
> >
> There is no cost to implement T.38. Open H.323 publicly suppo
On Thu, 2003-04-03 at 13:37, d hinton wrote:
> ok, that's what i've been asking all along. for cert numbers for FCC.
> as for cert process. it's easier then many folks have stated. in most places
> a mere statement of conformity by manufacturer will do (cost in US is ~$900)
> however testing is sti
Is there a block diagram or similar for the T400P?
What about programming information? What documentation
exists for it that a device driver writter might want?
I need a T1 interface for a non-telephony application.
All I need to do is get the raw bits into a computer
The application is this:
I have to disagree here. I send and receive faxes over IAX all the time
James
On Thu, 3 Apr 2003, Brian J. Schrock wrote:
> From what I have heard packetizing fax does not work well, does not
> matter if it is IAX or SIP. I think that was straight from digium tech
> support.
>
___
ok, that's what i've been asking all along. for cert numbers for FCC.
as for cert process. it's easier then many folks have stated. in most places
a mere statement of conformity by manufacturer will do (cost in US is ~$900)
however testing is still the way to go, as if any thing should happen you
w
I guess you mean the whole thing. It should be possible to port, and I
think I gave it a shot at compiling at some point, but I didn't spend
much time looking at why it didn't work. Surely, there's lots of very
small things, and probably a few big things that would need to be fixed
(most notably
As far as I can find out (till now that is) all it actually takes (since
2001) is conforming to the R&TTE
As in
http://www.radio.gov.uk/topics/conformity/document/rtte/rtteman/rtteman.htm
Annex 4 states that An application for an Opinion should be accompanied by
(amongst others):
Version o
Karl Putland wrote:
Well there's T.30 which is a store and forward mechanism, or T.38 which
is a realtime fax relay but they cost $ for licensing of the protocols.
There is no cost to implement T.38. Open H.323 publicly supports T.38.
Jeremy McNamara
_
Is it possible to give a false ringbakc on asterisk ?
--
Michael Bielicki
Managing Director
TAAN Consultants Ltd
http://www.global-gateway.net/
--
This correspondence is for the named person's use only. It may contain
confid
On Thursday 03 Apr 2003 16:58, Klaus-Peter Junghanns shaped the electrons to
say:
> Hi d hintion,
>
> hmmm...getting approvals for europe isnt that easy.
> because you get the approval for a combination of hardware
> and driver software, so when you change the driver you loose
> the approval.
that
On Thursday 03 April 2003 09:58 am, Drew Hamilton wrote:
> Any way to get the Asterisk to do the "broken dial tone" on phones
> that have voice mail in their mailbox? Similar to the phone
> company's built-in voice mail
/etc/asterisk/zapata.conf:
;
; Stutter dialtone support: If a mailbox is spec
Yah, just set "mailbox=" above the channel declaration.
Mark
On Thu, 3 Apr 2003, Drew Hamilton wrote:
> Any way to get the Asterisk to do the "broken dial tone" on phones that
> have voice mail in their mailbox? Similar to the phone company's
> built-in voice mail
>
> - awh
> _
On Thursday 03 April 2003 09:56 am, Jon Pounder wrote:
> I think the real solution is some piggy backed protocol that can be
> told this is fax information at one end, digitize the fax as if it
> were a faxmodem, stream it to the other end using a non-realtime
> protocol, and then initiate a fax ca
> On Thu, 2003-04-03 at 10:55, Mark Spencer wrote:
>> > 1. it was FCC, CA and CE certified (FCC and CA states no card is reg
>> with them as of last week)
For the T100P and T400P:
http://www.part68.org/tte_details.cfm?cicHistid=36439
http://www.part68.org/tte_details.cfm?cicHistid=36442
They wer
Unless things have simplified since I was last involved in European
approvals (which is quite a long time) things are worse than that. If
your factory has not previously produced approved telecoms products, you
probably need to pay for a factory inspection; each new protocol you
want to support
On Thu, 2003-04-03 at 08:56, Jon Pounder wrote:
> I think the real solution is some piggy backed protocol that can be told
> this is fax information at one end, digitize the fax as if it were a
> faxmodem, stream it to the other end using a non-realtime protocol, and
> then initiate a fax call a
The way I've seen it done is that the incoming fax signal is digitized and
compressed, then sent over the IP channel. It is done in real time. You
end up taking up 7k-14kbps instead of the 32/64kbps you'd use to pass high
enough audio quality to not irritate the modems.
Unfortunately, this takes
On Thu, 2003-04-03 at 10:55, Mark Spencer wrote:
> > 1. it was FCC, CA and CE certified (FCC and CA states no card is reg with
> > them as of last week)
>
> We do have FCC part 68 certification on both the T100P and T400P. When I
> get back I can look to find the numbers if you wish.
>
> I do no
You should be able to packetize your fax calls without any problems. We have hundreds
of fax and modem lines on MGCP IADs that connect to PRI gateways without problem,
although I haven't tried with IAX.
Be sure to use G.711 codec and disable echo cancellation, and if your network is
robust wit
sorry if you took it wrong, i was just repeating what FCC told me. not
trying to call you or your team liars. if it is listed, then i should be
able to find it in the part68 database, but it's not showing up.
dwayne
- Original Message -
From: "Mark Spencer" <[EMAIL PROTECTED]>
To: <[EMAIL P
On Thu, 2003-04-03 at 09:58, Drew Hamilton wrote:
> Any way to get the Asterisk to do the "broken dial tone" on phones that
> have voice mail in their mailbox? Similar to the phone company's
> built-in voice mail
If you are on a Zap device place a mailbox= before the
channel= line so asterisk kn
Some people run fax over IAX using ulaw codec on the local LAN.
Martin
On Thu, 3 Apr 2003, Brian J. Schrock wrote:
> From what I have heard packetizing fax does not work well, does not
> matter if it is IAX or SIP. I think that was straight from digium tech
> support.
>
> On Wednesday, April 2,
codecs/codec_mp3_d.c -> xing ...
On Thu, 3 Apr 2003, Michael Bielicki wrote:
> what does playback use ?
>
> On Thursday 03 Apr 2003 16:18, Martin Pycko shaped the electrons to say:
> > Can you use Playback instead ?
> > Playback doesn't use mpg123.
> >
> > regards
> > Martin
> >
> > On Thu, 3 Apr
I woudln't write that if it wouldn't support mp3.
On Thu, 3 Apr 2003, Tamas Levente wrote:
> And does playback support mp3?
> - Original Message -
> From: "Martin Pycko" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, April 03, 2003 5:58 PM
> Subject: Re: [Asterisk-Users]
Michiel Betel wrote:
Thats what I specifically asked... Should the whole system be approved (eg.
Computer, cards, software) or just the components. The answer I got back
from two agencies was that hardware approval under R&TTE should be
sufficient. Same as ISDN BRI card manufacturers do...
There a
This wheel has already been invented. Fax over IP usually uses a spec.
called T.38 It wouldn't make much sense to use anything else, since
implementing T.38 allows you to talk to a variety of existing FoIP kit.
Regards,
Steve
Jon Pounder wrote:
I think the real solution is some piggy backed pro
> 1. it was FCC, CA and CE certified (FCC and CA states no card is reg with
> them as of last week)
We do have FCC part 68 certification on both the T100P and T400P. When I
get back I can look to find the numbers if you wish.
I do not like your accusation that we are lying about our certificatio
what does playback use ?
On Thursday 03 Apr 2003 16:18, Martin Pycko shaped the electrons to say:
> Can you use Playback instead ?
> Playback doesn't use mpg123.
>
> regards
> Martin
>
> On Thu, 3 Apr 2003, Tamas Levente wrote:
> > Hey,
> > I've installed 0.59r mpg123 on a redhat box. I set the ex
hi kapejod
you said
"hmmm...getting approvals for europe isnt that easy."
the testing and cert filing is being handled by a national lab, i'm sure
they know what their doing. and if the drivers make the CE need to be
updated, at least it will be legal.
you also said
"oh yes, sure you can produce th
Thats what I specifically asked... Should the whole system be approved (eg.
Computer, cards, software) or just the components. The answer I got back
from two agencies was that hardware approval under R&TTE should be
sufficient. Same as ISDN BRI card manufacturers do...
-Original Message-
F
Nope not unless you want to port it.
--Mike
On Thursday, April 3, 2003, at 02:17 AM, Roy Sigurd Karlsbakk wrote:
hi
Can I use Asterisk with OS X?
roy
--
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801 3356
Computers are like air conditioners.
They s
Any way to get the Asterisk to do the "broken dial tone" on phones that
have voice mail in their mailbox? Similar to the phone company's
built-in voice mail
- awh
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And does playback support mp3?
- Original Message -
From: "Martin Pycko" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, April 03, 2003 5:58 PM
Subject: Re: [Asterisk-Users] MP3player problem
> Can you use Playback instead ?
> Playback doesn't use mpg123.
>
> regards
> Martin
I think the real solution is some piggy backed protocol that can be told
this is fax information at one end, digitize the fax as if it were a
faxmodem, stream it to the other end using a non-realtime protocol, and
then initiate a fax call at the other end and restream out the data, all
while po
Hi d hintion,
hmmm...getting approvals for europe isnt that easy.
because you get the approval for a combination of hardware
and driver software, so when you change the driver you loose
the approval.
oh yes, sure you can produce the cards and sell them cheaper,
but that doesnt take the developmen
From what I have heard packetizing fax does not work well, does not
matter if it is IAX or SIP. I think that was straight from digium tech
support.
On Wednesday, April 2, 2003, at 09:53 AM, John Harragin wrote:
Hi,
We are looking at consolidating our lines with PRI. This will allow the
elimina
Can you use Playback instead ?
Playback doesn't use mpg123.
regards
Martin
On Thu, 3 Apr 2003, Tamas Levente wrote:
> Hey,
> I've installed 0.59r mpg123 on a redhat box. I set the extension up for the
> mp3player. I called and it was playing the file back,but it was full of drops. like
> sound
you know, i asked diguim guys for this info and got nothing. so my company
decided to produce these cards for our own use and have them tested for FCC,
CA and CE certs
est. for testing we got back, range between $7,500 - $9,500 (USD) for all
three.
we are also surprised that the quote we got for p
You configure them as usual
zaptel.conf
fxoks=1-n #(n - how many cards you have)
Then you can just plug a single phone line to each of them
and then in zttool which one will go into OK from RED state.
regards
Martin
On Thu, 3 Apr 2003, Jim Archer wrote:
> Hi All...
>
> If I have more than 1 X1
I called around and got some rough quotes for R&TTE testing and
certification for europe. It seems to boil down to euro 2400,- per card to
be tested. They would also need the tech. doc and design from digium...
Any european users want to help? I'd like to be able to legally use the E1
cards
M
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or, via email, send a message with subject or body 'help' to
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Title: Message
NOTE: This one does not touch
localtime_r..
Here's a small patch
to logger.c i wrote which enables Asterisk to log to
syslog.
put a line like
below in your logger.conf:
syslog =>
notice,warning,error
and Asterisk
will write it's logging to /var/log/messages too..
Note t
Hey,
I've installed 0.59r mpg123 on a redhat box. I
set the extension up for the mp3player. I called and it was playing the file
back,but it was full of drops. like sound - silence - sound continued. I thought
let's try with the developer version of mpg123 cos other extensions like echo
tes
> > > No analog modems.
> > >
> > > If your ISDN adapter is supported properly, you can place that straight
> > > into a asterisk box. The analog line would need a X100P.
> > so to sum things up: one x100p for every analog line and one isdn4linux
> > adapter for every isdn line, right? i'd like t
Title: Message
Oops
just noticed my
patch removes the changes made in CVS to threadsafe the time routines... I'll
create a new diff which compares the latest trees...
Michiel
hi
Can I use Asterisk with OS X?
roy
--
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801 3356
Computers are like air conditioners.
They stop working when you open Windows.
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Title: Message
Here's a small patch
to logger.c i wrote which enables Asterisk to log to
syslog.
put a line like
below in your logger.conf:
syslog =>
notice,warning,error
and Asterisk
will write it's logging to /var/log/messages too..
Note that event-log
logging is not yet included. (sho
Lynx Automotive
Research & Development Area
In order to have a fully functional * box over a PRI E1 i know I have to buy an
E100P, but:
I've to configure an analog FAX with a direct number incoming from the
PRI.
Since all the telephony solution will be over
Hi All...
If I have more than 1 X100P card, how do I configure them so I know which
one is which?
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