Anybody looking to sell a Grandstream Budgetone?
Contact me off list if you have one you want to get rid of.
Thanks
-Brian (brc007)
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Klaus-Peter Junghanns wrote:
Hi,
make sure you have echo cancelation disabled on that zaptel
channel.
I tried that but no joy.
I've tried the gains at 0.8 and 1.5
I managed to get one fax to go out but it wouldn't repeat this feat!
Simon
--
Simon Faulkner - Dedicated Programmes
01538 303 900
Due to security reasons I want to run asterisk as a non root.
http://voip-info.org/tiki-index.php?page=Asterisk+non-root
This HOWTO works for great for me :)
F
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hi
I have set up a * box supporting 3 different companies but have some
questions regarding MOH. Can MOH support multiple context or classes.
Reason I ask each company would like to have different MOH sound files.
Is this possible?
yes, just specify multiple moh classes in musiconhold.conf
Sorry, It's not dyslexic being easy. The REAL area codes are
620 with a 221 prefix
316
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arnold Cavazos
Jr.
Sent: Saturday,
Adam -
I had a similar problem here in the UK using a Euro-ISDN PRI from BT.
The key was to add in the line pridialplan=unknown into zapata.conf.
Then it leapt into life in both directions. My files are below for your
information.
Rgds
Tim Robinson, Basingstoke UK
zaptel.conf
---
#
On Sun, 2004-02-15 at 00:46, mattf wrote:
I'd like to know why too.
I'm using a TE410P card in a dual Athlon XP system right now and it seems to
be playing nicely with the dual Athlons, should I worry about something
going wrong with my TE410P since it is basically the same card as the
A customer is looking to change to VOIP but he wants a local incoming #
where he lives. Anyone know a provider that offers them via SIP/IAX.
I'll be running Asterisk to run all the features.
Sincerely,
William Suffill
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Hello All,
Does anyone here have any experience with pingtel Xpressa hard phones? I am
considering buying a couple. Already have Snom200s, but want something with better CTI
and full duplex speakerphone.
Michael
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a Tyan Thunder K7X S2468, it only has one 3.3v PCI slot but it seems happy
with the te410p card.
MATT---
-Original Message-
From: Juan J. Sierralta P. [mailto:[EMAIL PROTECTED]
Sent: Sunday, February 15, 2004 10:22 AM
To: Asterisk Users
Subject: RE: [Asterisk-Users] TE405P and dual
Brian West wrote:
But CVS was alive the whole time! ;)
bkw
Um, no it wasn't.
For all practical purposes, *.digium.com was dead. Why? Because even
though there is a second cvs.digium.com out there on a different network,
the nameservers digium.com are both on the same network - the network
John Fraizer wrote:
There are several entities out there who will do secondary DNS for free.
You might want to look into that.
If you pull their NS entries from one of the root servers, you get:
digium.com. 172800 IN NS bos.nameserver.net.
digium.com. 172800
I was thinking about that... But here is my problem. We have 6 DID
lines. We have it set up that all three companies share all lines..
Based off of the DNIS states what AutoAttendant they hit. So if I were
to specify what channels the played certain MOH. Then that would mean
Company 1 would
Why people don't have al least some respect about regulations?
Sure that pridial=unknown solved that problem, but sadly you're overwriting
the main class of service indication in ISDN...
Unknown let to Class 5 switch manage (as the operator wish) understand
your messages.
The common sense shows
For the Stable cvs checkout, the asterisk.org site suggests:
To check out code from our STABLE 1.0 Branch CVS repository for Asterisk ONLY:
# cd /usr/src# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
# cvs login - the password is anoncvs.
# cvs checkout -r v1-0_stable asterisk
And without the secondaries knowing they're authoritative for the zone,
things don't work right.
John
David Coulson wrote:
John Fraizer wrote:
There are several entities out there who will do secondary DNS for
free. You might want to look into that.
If you pull their NS entries from one of
I've a Valcon V-2001A paging controller connected to an Adtran 750 FXS
port. The V-2001A looks like an FXS loop start extension.
When I call the extension, I can hear ringing tones and CallerID through
the speaker, but the paging controller doesn't answer--it continues to
ring. I also hear a
Hello list, I was going to buy this weekend a Wisip from
http://www.pulverinnovations.com/, but jeff got out of stock and he wont
have Wisip for the next 3 to 4 weeks. So I start searching for other
wifi phones because I was really upset about it and I found IPC5000 from
On Sun, Feb 15, 2004 at 12:18:03PM -0500, John Fraizer wrote:
There are several entities out there who will do secondary DNS for free.
I'll do secondary DNS if they want.
Tim
--
Tim Sailer Coastal Internet, Inc.
Network and Systems Operations PO Box
Hello,
In contrib/README.festival talks about a patch to extend the Extension
logic of Asterisk to allow put quotes and vars to the Festival commmand.
I need this since I want to use , to add a silence. Something like
this hello, how are you.
Now if I execute Festival('Hello, how
On Sun, 15 Feb 2004, John Fraizer wrote:
Brian West wrote:
But CVS was alive the whole time! ;)
bkw
Um, no it wasn't.
For all practical purposes, *.digium.com was dead. Why? Because even
though there is a second cvs.digium.com out there on a different network,
the
OK, time for this 1 day old Asterisk convert to start getting his feet wet
:)
I have just installed Asterisk on a Redhat box -- easy installation!
I am not using any analog interface hardware, instead am going to try to
test using my Vonage account.
S. The question is, how would I set
We don't support using Asterisk on your connection, but you're allowed to
use an Asterisk box if you can get it to work.
--
JustThe.net Internet New Media Services, Apple Valley, CA
Steven J. Sobol, Geek In Charge / 888.480.4NET (4638) / [EMAIL PROTECTED]
PGP: C57E 8B25 F994 D6D0 5F6B B961 EA08
Replacing the sip terminal for Vonage isn't possible. The terminal is locked
and will not allow access by the user to get the user/password info, and the
user/password handshaking is encrypted which prevents it from being spied
upon.
The only way this will work is if you plug the analog
On Sun, 15 Feb 2004 16:00:04 -0500, Tom Knox wrote:
I am not using any analog interface hardware, instead am going to try to
test using my Vonage account.
S. The question is, how would I set up Asterisk to replace my
Motorola VT100v voice terminal that Vonage provided me? I have been
Those phones look good, but, only have 10 milliwatt output.
Have you looked at these:
http://www.spectralink.com/products/nl-wts.html
100mw output.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Miguel
Cavazos
Sent: 15 February 2004 12:39
To:
Title: Mensaje
Hola,
ahi va la sección [es] para el indications.conf
[es]
description = Spain
ringcadence = 1500,3000
dial = 425
busy = 425/200,0/200
ring = 425/1500,0/3000
congestion =
425/200,0/200,425/200,0/200,425/200,0/600
callwaiting =
425/175,0/175,425/175,0/3500
dialrecall =
Tim Sailer wrote:
On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote:
I have been trying to start asterisk all night after a reboot
I keep getting this error scrolling up the screen
ouch: error while writing audio data broken pipe
when I go to another console there are 4
I had this problem with an old 16bit Sound Blaster Card.
Threw the card away and put in a cheap ?3.50 PCI card.
Works a dream now.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert
Boardman
Sent: 15 February 2004 23:20
To: [EMAIL PROTECTED]
Robert Boardman wrote:
Tim Sailer wrote:
On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote:
I have been trying to start asterisk all night after a reboot
I keep getting this error scrolling up the screen
ouch: error while writing audio data broken pipe
when I go to another
Hi,
Did HT-286 a good answer to put into my company and with both * and my old pbx use VOip and Normal telephones without change any kind of structure ?
Like with HT-286 can i just plug it using the RJ-45 part in the * and the RJ11 in my old pbx and have both working with my normal telephone ?
Hi all,
Is it possible to have multiple context= for user configuration in sip.conf?
Regards,
Antonio Rabena
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[EMAIL PROTECTED] wrote:
Hi all,
I received my shiny new TE405P on Friday, and after much fiddling and
assistance from the irc channel, I got a OK status (telco
reversed the TX/RX
and I wired it wrong).
Anyway, currently it works for inbound calls, but I can't
seem to dialout on
OK, I
Yep thats what it looks like :(
bkw
On Sun, 15 Feb 2004, John Fraizer wrote:
And without the secondaries knowing they're authoritative for the zone,
things don't work right.
John
David Coulson wrote:
John Fraizer wrote:
There are several entities out there who will do secondary
no but you can do this:
context=specialuser
in extensions.conf do this
[specialuser]
include=othercontext
include=yetanothercontext
On Mon, 16 Feb 2004, Antonio Rabena wrote:
Hi all,
Is it possible to have multiple context= for user configuration in sip.conf?
Regards,
Antonio
I don't know if anyone else has worked with Spectralink, but I tried to get some
demo units to test with a while back and I was really disapointed. At first they
claimed they were SIP complient. Then they sent me a contract for the demo.
They wouldn't send a demo unless I agreed to have them do an
Thanks Michael, VoicePulse does have local number, so I just provisioned
one :)
Now I am setting it up, no problem so far, the next question is. If I get 2
simultaneous calls on my inbound will one ring busy or will asterisk handle
this for me? I would like to be able to receive multiple
Hi all,
I'm having a hard time getting my calls to complete when creating call files
and putting them in the /var/spool/asterisk/outgoing directory.
Asterisk processes the file just fine, but the digits that enter my PSTN
switch are only 2 digits, not 10. For example:
Filename: 1.call (My call
[EMAIL PROTECTED] wrote:
However, I still can't dial any mobiles (all mobiles are 10 digit like
0402 xxx xxx) which I thought would have been the same as the above
local/std calls.
Also, I can't dial freecall numbers (1800 xxx xxx) etc.
If anyone has any hints on possible number formats
I agree, I think they would be useful too. Just don't know of anyway to
currently do it with *. Would also like to have a working intercom option.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Jeff Crews [EMAIL PROTECTED]:
I
Not really switch avice, but it you are rewiring anyway, ditch the cat3. A good
portion of wiring cost is the labor to pull the wire. Since you are doing it
anyone I would pull enough to handle the phones too.
We are shopping too, but for an enterprise wide solution (about 1000-1500 ports
WAN
That is also when I have seen a few crashes of 0.7.2, when doing lots of edits.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting John Fraizer [EMAIL PROTECTED]:
Geert Nijpels wrote:
I run 0.7.2 and have no crashes. Do you have
Hate to reply to my own post here, but it was a careless mistake on my part.
(My switch was giving imediate answer supervision on the inbound trunks)
Sorry for the trouble...
Thanks,
Seth
- Original Message -
From: Asterisk [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February
I purchased the Adit 600 to fix problems similar to what you are seeing and it
worked like a charm. Before investing in the Adit 600 though I would test to
make sure your pbx is actually sending the call supervision info for the call
supervision. If plug in a phone with a telco lit display or
Can anyone offer adivce for connecting * to a merlin legend ?
I'd like to use a t1 interface to connect the two, * will be used as a long
distance voip gateway in this scenario. Is this possible using a digium
t100p ?
Thanks,
Chris Clifton
___
On Sun, 15 Feb 2004, Chris Clifton wrote:
Can anyone offer adivce for connecting * to a merlin legend ?
I'd like to use a t1 interface to connect the two, * will be used as a long
distance voip gateway in this scenario. Is this possible using a digium
t100p ?
Is it possible? Absolutely.
You'll
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