imposed by the GPL?
--
Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx/~jtn/
GPG key fingerprint = 6272 5482 EDDD D0A3 FED2 262A FABB 599D FF67 6C9E
disclaimer: My opinions are my own. Don't bother my employer about them.
pgpDKA2WtCNcn.pgp
Description: PGP signature
business phone system.
Details of the project can be found here:
http://amp.voxbox.ca
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
403.244.8089
www.voxbox.ca
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and prevent you from using it. You still have the code. Who
says that this mythical MS-IAX solution would become a future de facto
standard? Your argument is weak. Let products stand on their own, or do you
perhaps believe that Asterisk could not stand on its own merit?
--
Jason T. Nelson [EMAIL
of the * box. im riping my hair out
on this one please help...
20:54:47.793156 192.168.1.15.51216 apollo.tftp: 28 RRQ
SEP00D0BA848162.cnf [tos 0x10]
Jason
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but that is where I would start.
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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as having lost anything; in fact, Apple
has returned many things to the community without being coerced as could
be said about GPL-licensed code.
--
Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx/~jtn/
GPG key fingerprint = 6272 5482 EDDD D0A3 FED2 262A FABB 599D FF67 6C9E
. But there was a highly publicized
alleged violation by Cisco/Linksys:
http://lkml.org/lkml/2003/6/7/164
To be honest I don't even know the end of that story (if it has
ended)... probably some bureaucratic snafu.
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
403.244.8089
www.voxbox.ca
dont think you understood the posters question.. he was asking if *
could be run over a ssh tunnel. not running admin commands via ssh
cli.
On Tue, 12 Oct 2004 23:05:42 -0400, Andrew Thompson
[EMAIL PROTECTED] wrote:
Christopher Jacob wrote:
Anyone ever set up Asterisk to use SSH
- Original Message -
From: Glenn Dalgliesh [EMAIL PROTECTED]
Subject: [Asterisk-Users] ValetParking
To: [EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
First Thanks to brian for work on valetpark it seems to work really well
I was working on
.
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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mine had the same FW on it when i started, i went to 6.3, and then 7.2
with no issues.
On Tue, 12 Oct 2004 16:43:10 -0500, Henry Devito [EMAIL PROTECTED] wrote:
The buffer is full on the 7960 because it keeps the old software along with
downloading the new software just incase the new
they would look into it (they
thought that it was iaxtel's fault). I waited a couple of weeks and
tried again to no avail. I gave up trying telesthetic/Iaxtel.
Telesthetic does work with FWD. I've tried this myself and it does
work even using FWD's IAX services.
Cheers,
Jason Stewart
Could someone please post the url for the conf? also mute your mic so
everyone can hear!!!
IAX2/[EMAIL PROTECTED]/4569
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Back in the office post-astricon. 1.0.0 running in the lab.
YIIIHAA!
THIS GUY rocks. Thanks to Mark for *, Steve and Olle for the conference
and to ALL community members. Everyone using * is contributing in one way
or another.
See y'all next year
Jason Kawakami
- Jason
On Jul 15, 2004, at 7:15 PM, chouck wrote:
I have a vt1000v laying around from someone who used to have service
with
vonage. Any idea how I could reconfigure it? The thing is locked
down, you
can change the webinterface and go to the secret page by typing in
192.168.102.1\admin.html
the Internet.
I cannot understand my problem with one way sound... what is wrong on my
configuration :((
As the IX66 is a sip aware router make sure you have no entries for
nat in your sip.conf, and let the ix66 deal with the nat, not * . I
hope this helps.
Jason
On Thu, 16 Sep 2004 20:22:57 +0900 (JST), Isamar Maia
[EMAIL PROTECTED] wrote:
Hi FOlks,
I am trying to setup remotely an AudioCodes Mediant 2000 MG Module 2 to
work with Asterisk through SIP or H323.
But since I don't the product manual, it's being a little hard.
Anybody would the manual
lookup DB.
Now that being said I am no expert on d-channel messaging so I can't really
answer the question on how/if we can pass the CALLERIDNAME across a private
d-channel connection between * and another PBX.
Jason Kawakami
www.optellabs.com
systems that support PRI
networking have some special d-channel signalling and it may just be a
question of building something in * that the AVAYA needs to see to
accomplish this.
Good Luck
Jason Kawakami
www.optellabs.com
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throught the IVR logic.
whatever, could be done either way.
Jason Kawakami
www.optellabs.com
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with a much smaller budget.
Jason Kawakami
www.optellabs.com
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are delivered with a higher
priority. Either custom queuing or bandwidth reservation or both will
make everyone's life better.
echo here
Jason Kawakami
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the signalling on the PRI to NI2 (or some other
signalling that gives name display I am in the US and that is what I always
have my carriers send me). That would give you name display from the
carrier.
Less work and the carrier probably doesn't charge more for it.
Jason Kawakami
www.optellabs.com
and dive in.
welcome to the brave new world
Jason Kawakami
www.optellabs.com
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to see what your
phone system is sending your voicemail system.
good luck
Jason Kawakami
www.optellabs.com
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with projects like this (for the
right price).
However you go, good luck
Jason Kawakami
www.optellabs.com
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a better way to
do
this as well. If so, please let me know.
Regards,
Paul
Good Luck
Jason Kawakami
www.optellabs.com
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On Thu, 26 Aug 2004 09:25:55 -0600, Andrew Elchuk
[EMAIL PROTECTED] wrote:
Is a sound card needed in order to playback some of the asterisk sounds
in /var/lib/asterisk/sounds when dialing out with an X100P? Thanks.
No Sound card is requied
___
the following line
fromdomain=sip.address.com
Regards
Jason
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highway works but MSN's are not
availiable and CLIP- Callerdisplay is not an option for the ISDN Line)
I have 8 MSN's Callerdisplay, Plus 2 analogue numbers all works great
through a fritz card.
Jason
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opinion ISDN is the way to go.
Jason
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running the CVS from yesterday. Any ideas?
In XTEN you need to turn off silence suppresion. AdvacedAudio
SettingsSilence SettingsTransmit Silence = YES
Jason
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only work for DDIs. If I have to use DDIs can anyone
recommend and active ISDN card which works with Asterisk and is
readily available in the UK.
I use a BT Speedway card and chan_capi under * with MSN's works fine no issues.
Jason
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if this is
the case?
Some sound during the conversation the card has detected as a busy
tone set busydetect=no in zapata.conf or increase the busycount=4 to a
higer value, if you need busy detection.
Jason
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to
another switch and does the recording but if it gives an indication that it
is recording and I can't change that I will have difficulty selling it to
the powers that be.
Does anyone know of a flag setting that makes 'Monitor' silent?
Jason Kawakami
On Tue, 17 Aug 2004 01:16:21 +0200, Patrick [EMAIL PROTECTED] wrote:
On Mon, 2004-08-16 at 22:13, Markus Engelbrecht wrote:
Hello,
so I decided to update to the latest CVS version of asterisk and of
chan_capi.
You are compiling the wrong version of chan_capi to get chan_capi to
work
but * is either missing
the first digit consistantly.
This seems to me to have something to do with start timers or inter-digit
dtmf timers or something.
I have even tied 2 * together each with t100 cards and have the same
problem.
Not sure how to proceed, any suggestions?
Jason Kawakami
of the phones that is
having trouble but still no luck.
I do not have access to any other images so if you want to send me one on or
off list that would be great.
Jason Kawakami
Jason Kawakami [EMAIL PROTECTED] wrote:
I have 4 7960's that I am trying to get working but 2 of them will not
update
for each phone? would be like cisco et
al to do something like that.
TIA
Jason Kawakami
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.
Assistance is greatly appreciated.
Michael
or am I confused on what you are trying to do?
Jason Kawakami
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that the customer wants to do differently than
the NEC's or Nortel's or the Avaya's are going to let them.
Jason Kawakami
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) the card no
longer seems to be able to detect ringing. I've tried upping the rxgain with
no noticable effect. Does anyone else have this piece of hardware working
with a TDM400P that could shed some light on this?
--
Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx/~jtn/
BOFH
is there something that needs to be set up to make the 'called' and
'callers' buttons work on this phone?
all i get is the backlight to switch on and off.
Jason Kawakami
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it should work fine on the PRI check the definity trunk form
and set send number to y
Jason
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,
The dial string from the X100p is the one that needs the t
eg
exten = s,1,Dial(SIP/2001SIP/2002,20,tr)
Also ensure in the phone that the DTMF mode is set to
DTMF RELAY inband(RFC2833)
DTMF Payload( 101)
Jason
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sure what the deal is.
ztcfg returns 24 channels configured but the same
notice at the bottom and when i launch * it fails because chan_zap cannot load
for some reason.
Jason Kawakami
having
endpoints within the system, they are connected to each other. I would
prefer not to use MeetMe because it will never be more than a one to one
connection. Anyone have any suggestions?
Best Regards,
Jason R. Park
E-Mail: [EMAIL PROTECTED
being used? Both cards are on
same IRQ as 2 other devices.
I don't know if that's what's causing your panics, but sharing an
interrupt on an X100P is a no-no. Sharing interrupts can cause all
sorts of headaches.
Jason
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wouldn't other than a faulty handset.
Any thoughts or experiences?
Jason Kawakami
Technical SalesOpen Telephony Labs,
LLC801.527.2284www.optellabs.com
and re download from cvs
Then
make clean
make install
Jason
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Classifier. It goes directly to the Televantage's default auto
Some more information on how the two systems are connected would help
are you using PRI, T1, Analogue etc...
Jason
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Jason
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include the area code.
Jason
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the port was opened you
could dial anything in the dialplan in *. down and dirty but also cheap.
good luck
jason kawakami
Open Telephony Labs, LLC
www.optellabs.com
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digit extensions
Jason
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On Mon, 2 Aug 2004 04:50:08 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On Mon, 2 Aug 2004, Jason Williams wrote:
On Sun, 1 Aug 2004 14:50:26 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I use Brian's Valet Parking on our system.
exten = 700,1
in the extensions.conf file.
Add this line to the cisco section
insecure=yes ; To match a peer based by IP address
only and not peer
and make sure the host=xxx.xxx.xxx.xxx is correct
Jason
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This is my mgcp.conf that works with my D-Link 1120M
I hope that it helps.
-Jason
[general]
;port = 2427
;bindaddr = 0.0.0.0
[dvg-1120m]
host = 10.251.251.253
context = internal-phones
canreinvite = no
callwaiting=yes
cancallforward=yes
threewaycalling=yes
transfer=yes
I do that. But when I play MusicOnHold the music is played slowly! I don´t know
why... but is how the bitrate is playing with a different number.
Make sure you are running mpg123 0.59r and no other version
Jason
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| configured Default then I was off to the races.
Or I think you might be able to right-click on the interface (which
doesn't work under Wine, last time I checked; it locks up the interface)
and choose the account you want to use.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http
port
produces a dialtone on the phone). I've even tried switching the line polarity
and got no different results. This worked fine under the 0.4 drivers as I was
using the box quite heavily for testing purposes.
--
Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx/~jtn/
BOFH
.
Thanks -Jason
--
Jason A. Kates ([EMAIL PROTECTED])
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.
that single card would give you up to 96 ports of FX(x), much more flexible
etc.
good luck
Jason Kawakami
Open Telephony Labs
I am looking to do this as cheap as possible since the main reason we want
to upgrade is to take advantage of the VOIP functionality in Asterisk. We
also want to use IAX
I've tried setting nat=yes in places, externip, et al with no success ..
even though the code I was running from back then worked without that.
Some of the options in sip.conf have changed look at the samples in
src/asterisk/configs/sip.conf.samples
Regards
Jason
don't
think I just want setup so no login is required. Please help
Check out the dial command
Show application dial
dial(device1device2device3)
Jason
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- Original Message -
From: Steve McMahon [EMAIL PROTECTED]
Date: Fri, 23 Jul 2004 01:12:26 -0700
Subject: [Asterisk-Users] Cisco 12sp firmware... Anyone got it???
To: [EMAIL PROTECTED]
Looking for firmware (anything) for the 12sp model phones. Anyone got
it drop me a line!
transferring, so a single pound transfer option is unacceptable.
Where did the code go? How can I apply the doublehash patch? I know there
are several other people out there that go through what I do every time we
res_parking has become res_features so look there somewhere
Jason
Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
Reply-To: [EMAIL PROTECTED]
I haven't really gotten too far into this, but I was wondering just what
'features' of the NEC phones (DTH-16D-1(BK)TEL) I'll be able to work
with from *? I'm currently getting some
chan_capi.so=yes
Jason
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On Thu, 22 Jul 2004 12:10:25 +0200, Diego Ercolani
[EMAIL PROTECTED] wrote:
Il 10:03, giovedì 22 luglio 2004, Jason Williams ha scritto:
ensure you have the following in the [global] section
[global]
chan_modem.so=yes
chan_capi.so=yes
sorry, why do you need chan_modem? I don't
On Thu, 22 Jul 2004 14:54:29 +0100, Steve Hanselman
[EMAIL PROTECTED] wrote:
Yes, you'd have a dialplan entry that set a value in the database, then
acted upon that.
You'd probably want some nice voice prompts
The system is currently in [Day/Night/Holiday] mode, press 1 to set to day,
2
-Original Message-
I just want to clarify a few things. I have about 100 Toshiba digital
phones, 4 ports on the voicemail, and 24 phone lines. Not all of the
lines are POTS lines. I think 8 of the lines are Direct Inbound Dial
(DID). Due to a decrease in call volume, I am most likely
paying for the Vonage service (it will forward to my
cell in the event that the PBX was unavailable) I really would like to
keep my costs down by finding the lowest priced VSP out there. I don't
make calls to Europe or Canada.
Thanks in advance...
Jason
-Original Message-
From: [EMAIL
Greg, Chris, and Jay,
Thanks! You've given me plenty of info to digest. I really appreciate
the responses. Apologies if my list manners aren't up to snuff!
Thanks again,
Jason
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a configuration error somewhere it looks like the IX66 is trying
to authorise the clients, and no * have you set the IX66 to forward
all sip requests for your domain to * ?
Jason
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.
Are the
Intel's any better than the Digium? Any assistance would be greatly
appreciated.
Good Luck!
Jason Kawakami
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normally wait about a second after I pick up the phone until I hear a
very small click. I think that might be the end of the training period.
~ Then I proceed with my introduction. It seems to work quite well.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com
On Tue, 20 Jul 2004 10:49:51 -0400, Seth Remington
[EMAIL PROTECTED] wrote:
You are probably having a problem with parking being renamed to
features. Try a make clean then a make install. If that doesn't work
then delete the res_parking.so module from /usr/lib/asterisk/modules/.
You may need
into some sort of a codec mismatch or something because it's not working
right. The SIP client is a SPA-3000.
Phoneboy
IAXcomm use gsm only that may help
Jason
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to use # transfer which will mean you
will not be able to dial # into ivr's.
Search on wiki for # transfer
Regards
Jason
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,Dial(SIP/Nick)
exten = 21062,1,Dial(SIP/Sharon)
[internal]
exten = 310,1,Dial,Zap/2
include = sip ; allow internal to dial sip phone
Try those changes and see how you get on
Jason
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think I'll need the U30, but I'm not entirely sure.
The U20 will be fine. The U30 adds MF receivers for Feature Group D/E911.
The T-1 is just set up as EM tie line.
Good Luck
Jason Kawakami
Open Telephony Labs, LLC
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Date: Mon, 19 Jul 2004 14:54:44 -0500
From: Christopher L. Wade [EMAIL PROTECTED]
Organization: Unistar-Sparco Computers, Inc.
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
Reply-To: [EMAIL PROTECTED]
Would the TLI(2)-U10 ETU work as well?
That
I am trying to setup a simple pstn gateway using Asterisk and a X100p card.
I have got everything installed using Redhat 9 and am able to load Asterisk.
I also configured sip and I am able to connect to the asterisk gateway with
Xlite on the windows side.
I am able to dial 1000 and get the
[EMAIL PROTECTED] | FWD [EMAIL PROTECTED] | IAXTEL 1700 337 68
48
On Jul 18, 2004, at 5:12 PM, Jason Armentrout wrote:
1 channels configured.
It appears that I have the driver loaded correctly.
I edited the sample extensions.conf and changed the varible trunk to
zap/1
Attached
94341321 or 4341321 I just get a 404 error in Xlite.
What am I doing wrong? Any help would be appreciated.
Hey Jason
In your extensions.conf, the [default] context only has the [demo]
context included which provides no outbound dialing. Try adding an
'include =' line to your default
the
zaptel from picking up the line until the sip phone actully answers the call.
This way I could answer the phone either locally on a regular analog handset or
through the sip phone.
The way it is now, it only rings my phones in the house 1 time.
Jason
Quoting Marty Mastera [EMAIL PROTECTED
Maybe another nice feature might be for the other end of this problem...
Menu options while you are on hold to change the crappy music or mute it.
Also an option to punch in a callback number and have the company ring
your phone when it is your turn to have the call answered.
- Jason
I have
I started having problems with their IAX termination service last night. I
couldn't make any outbound calls but could receive inbound. I made the
changes to the configs that were e-mailed to me and now it is working
fine.
- Jason
I'm a bit displeased at the way this happened. I received
device available from
www.virbiage.com. It might work with more than just their FireFly
client, especially if it shows up as an audio device under Linux.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux
be careful.
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andres
Sent: 12 July 2004 08:54 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
So isn't this the problem * has? The first client
correctly.
Jason
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, seeing as the only files modified are
app_queue.c and chan_agent.c
I cannot vouch for this, as I've never used shady dial, but I will be
and I'll be sure to give my review of it in the near future.
Cheers,
Jason
P.S. Please do not reply to an existing message; this wastes bandwidth
and screws up
give you an idea. the syntax might not be perfect but it should
get you going down the path.
when in doubt, check the wiki www.voip-info.org or google for something
like IAX trunking. you are not the first person to do this.
good luck
Jason Kawakami
Try using $AGI-stream_file(filename)
There are built-in AGI commands - you don't have to use exec for all
commands.
Hope this helps
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bruce Marler
Sent: 07 July 2004 08:43 AM
To: [EMAIL
/RED - YEL/REC - Red/REC - OK. Eventually settles into RED.
Looks like you have a card problem a loop back to yhe T100P should go
green in about 3 seconds like the channel bank.
Regards
Jason
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=dynamic
canreinvite= no
qualify=200
dtmfmode=inband
defaultip=XXX.XXX.XXX.XXX
callerid=MediaTrix Port 1 602
mailbox=602
I would remove the mailbox line as the voicemail notifications may
well be causing the problem
Jason
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version of asterisk are you using a fix went in last week for
spurious call detections on the TDM400P. I should try downloading the
latest zaptel drivers and asterisk code.
Jason
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this :
-- SIP/10.10.1.1-babc is ringing
The codec's should work fine together I think you have a NAT traversal problem
Jason
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On Wed, 7 Jul 2004 10:35:56 +0200, Xavier Olivella [EMAIL PROTECTED] wrote:
When installing asterisk, I follow de Getting Started manual and i get the
following error while compilig asterisk:
if [ -d CVS ] ! [ -f .version ]; then echo CVS-HEAD-08/06/04-09:53:57
.version; fi
bison
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