go back to 1.35
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
more explanation Have you
selected an outbound line yet ? there is no Dial line in this
extensions.conf I am confused but it could just be me.
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
On 06/07/04 15:17 -0400, Joe Baptista wrote:
-- Forwarded message --
Date: Wed, 07 Jul 2004 00:31:21 -0400
From: Declan McCullagh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Politech] House bill exports analog phone regs to VoIP
On Wed, 07 Jul 2004 16:50:18 +0300, Damian Minkov [EMAIL PROTECTED] wrote:
I have * box on machine with external ip address and internal one
I'm tring to register to it from to machines - one from innternet (
everything is ok - in sip.conf nat=yes)\
and the other one is in the internal network
checkout
http://www.voip-info.org/wiki-Asterisk+consultants and find a
consultant in your area to help you.
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
, Asterisk probably
isn't the best choice for you.
Good luck
Jason Kawakami
Optellabs
Any help is greatly appreciated.
Thanks!
-Mike Wagner
MCCESC
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo
I had a simular problem. It may not be dialing because it detects another
phone has picked up the line and the voltage has dropped. I tried the same
test. It turns out it WAS dialing but my txgain was at 15! and the DTMF
was gettin distorted
set your txgain=0 and your rxgain=0
I've come across
,
IAX2/demo:[EMAIL PROTECTED]/s) in new stack
-- Called demo:[EMAIL PROTECTED]/s
Your iax dial line is incorrect
it is sending s as the extension
can you post the relevant section form your extensions.conf
Jason
___
Asterisk-Users mailing list
[EMAIL
be to build a SIP gateway with * and tie to the
Axxess with T-1.
Have Fun!
Jason Kawakami
optellabs
Message: 12
Date: Thu, 01 Jul 2004 23:43:33 -0400
From: Vasyl Rublyov [EMAIL PROTECTED]
Organization: IonIdea, Inc.
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Inter-Tel Eclipse2 (IP PhonePlus
expensive.
You do also get some additional software that is pretty cool but not for the
cost and nothing that a sharp coder couldn't bang out with a bit of effort.
Jason Kawakami
optellabs
Message: 8
Subject: RE: [Asterisk-Users] Inter-Tel Eclipse2 (IP PhonePlus)
Date: Fri, 2 Jul 2004 09:40:49
You can connect two PRI devices together using a T1 crossover cable.
from the zaptel sample config:
# dacs: The zaptel driver cross connects the channels starting at
# the channel number listed at the end, after a colon
# dacsrbs : The zaptel driver cross connects the
could tweak the DTMF recognition but that
doesn't look possible without recompiling dsp.c every time I make a change
to test it.
- Jason Garland
FWD# 273307
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo
could tweak the DTMF recognition but that
doesn't look possible without recompiling dsp.c every time I make a change
to test it.
- Jason Garland
FWD# 273307
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo
for division 2
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
At 16:47 25/06/2004 -0400, you wrote:
High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
Version 0.59s-mh4 (2000/Oct/27).
Looks a bit old to me... I'll try to install a newer release.
You need version r this is the only one that works well with asterisk
sense to anyone?
FWD only supports ULAW comment out the line allow=GSM in the general
section of the iax.conf
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
person is already listening
This will allow only one call to
use the resource music on hold.
Jason
At 14:48 25/06/2004 +0200, you wrote:
Hi!
FWD only supports ULAW comment out the line allow=GSM in the general
section of the iax.conf
Nonsense - FWD *does* permit the use of GSM.
Cheers, Philipp
Not in iax only with sip
Jason
___
Asterisk-Users
number to them
they will send out your name with the number but otherwise I have had little
success.
I have heard of working with a local/regional ps or carrier 911 coordinator
to have this fixed but have had no experience with it.
Jason Kawakami
is so low that it's useless. However, if you plug a handset
directly into the line coming from the CO, it's at a decent volume
level, etc. No echo, of course.
--
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
--
This message has been scanned for viruses
At 13:34 22/06/2004 +0300, you wrote:
I have problem compiling it
chan_zap.c: In function `zt_get_history':
chan_zap.c:768: storage size of `hist' isn't known
chan_zap.c:771: `ZT_GET_HISTORY' undeclared (first use in this function)
chan_zap.c:771: (Each
At 14:39 22/06/2004 +0300, you wrote:
I've compiled and run it but no effect.
Then i noticed that there is warning when i run asterisk
Jun 22 14:37:47 WARNING[16384]: chan_zap.c:8762 setup_zap: Ignoring ukcallerid
Make sure you have the correct switch in zapata.conf
callerid=uk
Regards
Jason
At 10:30 22/06/2004 -0300, you wrote:
Hi!
callerid=br exists?
miklos
Not unless you write the code
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit
| the root cause is still waiting on someone with the problem _and_ the
| programming skills.
I absolutely second this notion.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG
with the AGGRESSIVE_SUPPRESSOR switch (same file).
Whoah. I've used the AGGRESSIVE_SUPPRESSOR switch, but what does
CONFIG_ZAPTEL_MMX do for you?
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux
At 18:58 21/06/2004 +1000, you wrote:
On Sat, 2004-06-19 at 19:27, Storer, Darren wrote:
Hi Kevin,
KW By the way, it's useful to map 911 and 112 onto your 999
KW route for the benefit of foreigners who don't know any better.
Well, while you are at it, you might as well add-in 000, because
under Intimate. If I have the mixer settings too high for the master
volume, I get an awful feedback loop going. Don't know if the iPAQ has
any sort of builtin feedback suppression circuitry or not.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP
, but it's all source code.
He says he will release the sourcecode when he gets to a stable
working release.
Do you think your QtIAX client will run on a 206MHz StrongARM processor?
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE
/7000
Extension: 7000
Context: local
Priority: 1
ActionID: 1
The error I get in the CLI is Jun 17 08:18:36 WARNING[180236]:
chan_iax2.c:4534 socket_read: Call rejected by #IP: No compatible Codecs
Does anyone have any ideas.
Thanks in advance
Jason
Hi Adam
Done all that but still the same problem.
Do you have any other ideas?
Cheers
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart
Sent: 17 June 2004 08:29 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX2
it ???.
Any ideas, anyone
Thanks again Adam for the help
Cheers
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart
Sent: 17 June 2004 09:19 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX2 no compatible codecs
iax2 debug
At 16:49 16/06/2004 -0400, Eric wrote:
I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04).
These two boxes talk to eachother via sip, not iax. Since the upgrade, I
get the error Failed to authenticate on INVITE trying to make calls to/from
either box. Removing the secret
will start playing a congestion tone (not from
asterisk), and cease tx/rx audio. After this happens, a 'show channels'
will show that the call is still active.
Make sure you have VAD turned off and silence suppression turned off * may
not be
getting a continuous RTP stream
Jason
gotten iaxComm
to compile and work on an iPAQ H3670 running GPE/Familiar Linux in the
past. ZiaxPhone is supposed to run on the Zaurus or Opie.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment
file. The next time asterisk is
restarted
| it won't try to load the old iax stuff.
Which will promptly cause gnophone to quit working. :)
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jason A. Pattie wrote:
| You might look into using iaxComm or ZiaxPhone (available from
| http://iaxclient.sourceforge.net/). I have successfully gotten iaxComm
| to compile and work on an iPAQ H3670 running GPE/Familiar Linux in the
| past
my home E-mail again.
Thanks -Jason
--
Jason A. Kates ([EMAIL PROTECTED])
Fax:208-975-1514
This is an issues with DTMF clamping, you need to use chan_capi to get DTMF
working correctly.
Jason
At 18:30 15/06/2004 +1000, you wrote:
Hello all,
This afternoon I had a BRI line installed by Telstra (our telco in
Australia). I'm using an Eicon Diva 2.02 PCI ISDN card with the isdn4linux
debug in the CLI and see if you can get more
information on the error.
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
Wolfgang,
You need to use ccs not cas
I would change your timing options to these:-
-/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
span=2,2,0,ccs,hdb3,crc4
bchan=1-15
bchan=17-31
bchan=32-46
bchan=48-62
dchan=16
dchan=47
That should work
At 12:33 14/06/2004 +0200,
:
`__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__'
undeclared here (not in a function)
make: *** [chan_capi.o] Error 1
version 0.3.3 has been running fine without issues
Can any one assist (I'm sticking with 0.3.3 for now)
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com
to work reliably.
Regards
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Any chance of publishing source code as this is a good starting point for
many applications.
At 16:54 08/06/2004 -0700, you wrote:
I just uploaded a beta CallerID program.
It talks through the Asterisk Manager .
Pretty self expanatory for setup and configure.
Please Let me know what you think.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Adam Hart wrote:
| Jason A. Pattie wrote:
|
| |
| | One workaround is to use Firefly, but that may not be for everyone?
|
| True. I almost got it working under Wine, though. Kept dumping files
| into C:\. Probably just means I don't have
It works without a problem for me, but that does not help you.
Jason
At 17:29 09/06/2004 +0100, you wrote:
Hi everyone,
I've just got my X100P card installed and working but there seems to be an
issue with hang-up supervision.
If I stuff a call out over the X100P card onto the PSTN that's fine
I recommend a PRI E1 link into the Hicom from *
Jason
At 16:05 08/06/2004 +0200, you wrote:
Hi * :-)
I found in the online WiKi docs some information on how to integrate
Asterisk with old PBX...
http://www.voip-info.org/wiki-Asterisk+legacy+integration
...but I couldn't find anything
fixme:thread:NtQueryInformationThread info class 9 not supported yet
Oh well. It was worth a shot. At least part of the interface shows up
on the screen before Wine bombs.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU
That sounds like a timing issue, and that slips are occuring, Ensure all
cables are good, reolace if necessary and ensure you are timing off of the
network and not providing your own timing.
Jason
At 23:38 04/06/2004 -0700, you wrote:
I'm the one who posted the original question, and yes, I
It cannot be done in * so in your pbx set the first number to divert to the
second on busy and the other way and not ring both numbers. that will
resolve your issue
At 09:14 01/06/2004 -0300, you wrote:
Hi all,
Anyone know how put my X101P cards to answer at different ring times ?
Like
config file you are using the hicom as the second timing
source make sure the hicom is not clocking off of this line
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
The R button can do a hook flash when configured correctly.
Jason
At 20:24 30/05/2004 +0100, you wrote:
thanks for the reply, i thought it may be a stupid question but if i hit
either hook buttons i do not get any result when in a call. if i press
the hangup button it hangs up, press the pick
At 19:49 01/06/2004 +0900, you wrote:
I'm installing E100P for isdn pri line.
My configuration are like this.
zaptel.conf
===
span=1,0,0,ccs,hdb3,crc4
loadzone = us
defaultzone=us
bchan=1-15
dchan=16
bchan=17-31
zapata.conf
would need to be
written.
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi All
Does anyone know how I can get more information about an incoming SIP call
from a SIP proxy. Like FWD or any other SER proxy. My * box shows the
channel name as:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Adam Goryachev
Sent: 26 May
)
Ie they are quite random. I would far rather have something like
SIP/fwd.pulver.comXX
Does anyone have any suggestions
Thankls in advance
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Adam Goryachev
Sent: 26 May 2004 10:22 AM
or if it always trashed the other box when I
tried to bring * down with TDMoE in place.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
At 16:00 25/05/2004 +0530, you wrote:
But the issue is, it was hanging on 'Answer' application and
throwing out 'Unable to Spawn Extension (vpk, s, 1) . .
Do you have an extension s in context vpk ?
Can you provide the relevant section from the extensions.conf
Jason
analog handsets into those
devices that convert the analog extension into either a SIP or IAX phone?
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http
i always use the Goto application. seems to work quite well for testing
those s extensions.
exten = 2500,1,Goto(context,s,1)
will take you to step 1 in the s extension in whatever context.
Jason Kawakami
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday
to the IAX2
extension.
any ideas?
Jason Kawakami
This is normal for all VoIP communication there is nothing to wory about
and the lag is not heard in normal use.
Jason
At 13:50 20/05/2004 +0530, you wrote:
Hi,
IF i use a sip softphone or a iax softphone
with asterisk, i get a lag of about 1 second.
The two phones were on 2 different pc's near
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Anon wrote:
| On Thursday 13 May 2004 11:57 pm, Jason A. Pattie wrote:
|
|-BEGIN PGP SIGNED MESSAGE-
|Hash: SHA1
|
|Steven Critchfield wrote:
|| On Tue, 2004-05-11 at 17:33, Jason A. Pattie wrote:
||Our problem ended up not being with Asterisk
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jason Williams wrote:
| This is normal for all VoIP communication there is nothing to wory about
| and the lag is not heard in normal use.
|
| Jason
|
| At 13:50 20/05/2004 +0530, you wrote:
|
| Hi,
| IF i use a sip softphone or a iax softphone
That is the way it works over one zap channel, to keep * in the call it
would need to dial out on anoher line and that would then use an additional
zap interface and tie it up for the duration of the call.
Jason
At 20:30 18/05/2004 -0300, you wrote:
Yes, I've tried with SendDTMF, and it works
local servers nightly. I cannot wait to try this out :)
--
Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx/~jtn/
BOFH Extraordiaire Sysadmin Ombudsman GPG key 0xFF676C9E
GPG key fingerprint = 6272 5482 EDDD D0A3 FED2 262A FABB 599D FF67 6C9E
disclaimer: My
/indications stuff.
Help? Thanks in advance.
Jason Kawakami
for expansion in the future, if needed.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
iD8DBQFAqTDQuYsUrHkpYtARAkH4AKCBVA3Jhi3TFgWQkcCNcJvu
Are you running the latest cvs-head on both boxes if not update and try again
Jason
At 05:50 16/05/2004 -0400, you wrote:
Let me run this by the group, my inter-IAX connections are extremely
pop-hickup,missing syllable type of affairs.
Played with jitter buffers, card levels (this is totally
it.
Anything insecure is not the fault of the service itself, but the
fault of the expert in charge of making it secure. There was one
good source in the article, which was the MCI tech. He's not too
worried, since MCI encrypts all of their voip traffic.
Jason
in the console log
Unknown RTP codec 109 received
Is this a bug in chan_capi ?
Jason Williams
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
Just read my own message and it is not clear.
An inbound call to * via a BRI
when the caller presses # asterisk detects # and plays transfer message
when the call recipient (on SIP) presses # the tone is sent out over the
BRI rather than detected by *
and the error displayed as below.
Jason
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Steven Critchfield wrote:
| On Tue, 2004-05-11 at 17:33, Jason A. Pattie wrote:
|
|
|Our problem ended up not being with Asterisk or Digium hardware. It was
|the analog cordless phone. We simply have to live with it. What
|happens is whenever
can you assist. or give me a date that this package will compile against
Regards
Jason WIlliams
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
Thanks that resolved my compile issues
At 19:24 12/05/2004 -0400, you wrote:
Jason,
Lucky I had the solution in front of me.
Read here:
http://lists.digium.com/pipermail/asterisk-users/2004-April/044125.html
You basically need to run a patch against chan capi.
Regards,
Kimble Young
wanted to get it
out there in case anyone else runs into it, too.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
drive
space on my first Asterisk system before I realized what was happening
(2.5GB free space ... gone). I didn't see a need to keep the files, so
I've just been deleting them, but I'd like to not have them recorded in
the first place (or is this something that isn't legal?).
Thanks.
- --
Jason
Hi all
Does anyone have any experience with an E1 Channel bank connected to
Asterisk. I know form an earlier post that Orion telecom and Valiant Telecom
make them. But does anyone know if/how well they work with Asterisk.
Thanks in advance
Jason
-Original Message-
From: [EMAIL
Hi all
Does anyone have any experience with an E1 Channel bank connected to
Asterisk. I know form an earlier post that Orion telecom and Valiant Telecom
make them. But does anyone know if/how well they work with Asterisk.
Thanks in advance
Jason
try setting immediate=no
for that span
Jason
At 18:12 07/05/2004 +0100, you wrote:
Hi all
OK this may sound like a good one but maybe someone can tell me.
Simple context is - I want to unplug my existing conventional PBX from the
Telco and place * with it's TE410P in between.
Now
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Eric Wieling wrote:
| Allow ULAW or ALAW, not both, at least for trying to solve a problem.
What is the difference between these codecs? Which is better?
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN
There are later versions of the firmware that work better
Available from pulver and other places
Jason
At 03:07 25/04/2004 -0500, you wrote:
I do have a WISIP and it doesnt give me any problems im all day long on
the street using it. You cant talk of a phone you havent even touch
Miguel
On Fri
%3DcountryUK%7CcountryGB
HTH,
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Is anyone using Asterisk with a 3rd party voicemail system perhaps one that
uses the SMDI interface?
Thanks,
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
On 21/04/04 08:37 -0500, Sean Bruton wrote:
I am having some difficulty getting a T100P card to work with my PRI.
When I attempt to make an outbound call via:
exten = 1004,1,Dial(Zap/g1/NPANXX)
I see the following on the asterisk console:
-- Executing Dial(SIP/sbruton-b8ce,
It is the same product as listed below, with a different firmware
The firmware does exhibit similar problems
Jason
At 12:42 15/04/2004 +0100, you wrote:
We are currently integration testing the wireless Zyxel Prestige 2000W,
and if all goes well we'll have it for sale in 2 weeks. Has anyone
the new machine has an nForce2 chipset and
likes to assign the same IRQ to lots of different things.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http
I can't seem to find where I am supposed to create the config file, nor
do I know what the default admin password is..
Any suggestions?
Have a look at the readme in the zip file, specifically the bit about the
xml config file. Haven't made it work yet myself but it is probably a good
place to
G'Day,
I have a bit of FireFly problem that hopefully someone has seen before.
What happens is if I make to or receive a call from the FireFly network
the call will connect successfully. However, around 10 seconds after I
answer the call I am disconnected. The weird thing is same thing happens
Felix,
how can I checkout ztdummy?
Thank for you help.
Checkout of cvs the zaptel source then follow these instructions:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy
JR
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
JORA ROME wrote:
I wan work * whith SIP Communicator, it is posible?, what is
configurations? who can helpme?
Thanks
I couldn't get it to work either, circa early February. I had some
correspondence with the developer and sent him logs, etc. but nothing
ever came of it. He did say there were
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Doug Meredith wrote:
| Jason A. Pattie [EMAIL PROTECTED] wrote:
|
|
|Is there any possibility to remove the turnaround leg or whatever its
|called at the X100P? I'm just thinking of a scenario where none of the
|outgoing signal is ever introduced
Scott Laird wrote:
On Mar 31, 2004, at 8:06 AM, Mark Buckaway wrote:
In setting up Asterisk, I'm looking to dump my current phone system
(Nortel Venture). I presently have three POTS lines.
I would use a VOIP provider, but now are presently available in the
Toronto, ON, CANADA area that
the incoming circuit.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
iD8DBQFAavwMuYsUrHkpYtARAuZJAJ0ZmyHHMEQRPVa8uc9HYInjStofJwCfeOcn
Hi Chris,
Chris HARIGA wrote:
If you pay 8 USD for 1 year support you can download the image :)
Best regards,
Someone mentioned a while ago that even if you have a support contract
for the SIP image this doesn't mean you have a license to use the software.
JR
anyone already setup such a thing? If not, I could lend resources towards
such a project.
--
Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx/~jtn/
BOFH Extraordiaire Sysadmin Ombudsman GPG key 0xFF676C9E
GPG key fingerprint = 6272 5482 EDDD D0A3 FED2 262A
[EMAIL PROTECTED] wrote:
Another topic of interest is securing the box itself. Does a firewall
(hardware outside of the box or a linux based firewall) suffice the need?
Depends what you are protecting against. If you want to assume some services are
exploitable, you could try to break some
Back in February I found * pinning the CPU (Slackware 9.1, Feb CVS). I
ran a strace and found that it was looping on this:
-begin-
write(1, \nUse STOP NOW to shutdown Asteri..., 35) = -1 EIO
(Input/output erro
r)
write(1, *CLI , 6) = -1 EIO (Input/output error)
read(0, , 1)
. A virus could just match up addresses randomly
from the user's address book and files laying around on the hard
drive.
Cheers,
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
On 19/03/04 14:11 +1100, Master Abi wrote:
Hi,
G726-32 codec from beta firmware 1.0.4.54 now works fine with *. Tested
on BT101 and HT286 over a 64K DSL line. Some progress but iLBC still has
not surfaced.
Great news. This fw update breaks NTP sync in the phone for me, but
your milage may
On Fri, 19 Mar 2004 13:06:16 -0600, asterisk-users-request wrote
Send Asterisk-Users mailing list submissions to
[EMAIL PROTECTED]
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with
1201 - 1300 of 1377 matches
Mail list logo
