Hi List,
I've just built a new * box (slackware 9.1) and I get
no sound from a Playback(tt-weasels) command.
I've got other slack9.1 boxes running.
* Version is v1.0 stable
exten = 213,1,Answer
exten = 213,2,Playback(tt-weasels)
exten = 213,3,Playback(tt-weasels)
exten = 213,4,Hangup
when
Hi List,
Two boxes
A has a PRI
B terminates SIP devices
A --IAX-- B
Both on the same switch, same IP network.
Call from PSTN to A gets pushed via IAX to B - Sip device
with no problems.
Call from Sip device - B via IAX - A - PSTN
will drop exactly 5 seconds after the call is
Hi List,
I'm trying to get the DIALED NUMBER on a inbound
800 call.
I need to know what number the calling party called
so that I can route the call properly.
I really don't want to burn a DID per WATTS line
so that I can route on the DID number.
Any pointers??
john brown
The original message isn't copied because its HTML encoded.
Basicly the author wanted to know if he could use a HT-286 (Grandstream)
to bypass the PBX, generate busy, answer calls, etc.
The Grandstream HT-286, and the Sipura SPA-2000 are both
ATA FXS based devices.
ATA == Analog Telephone
From a New Mexico perspective,
When you order a PRI from a CLEC they typically will dump
your CLID info and replace it with the main number on the
span. You can request that they not do this and that they
pass the CLID thru.
We have been working with some of the local PSAP's here,
CLEC's and
rumor has it that new firmware will let you adjust the
ringer voltage on HT-286 / HT-486
On Mon, Feb 23, 2004 at 09:41:24AM +0100, Nicolas Bougues wrote:
Dear all,
My GS ATA-286, which otherwise work well, seem to be unable to ring a
fax (or at least, some kind of fax). The fax basically
Hi List,
how does one route calls in extensions.conf via DNIS ??
I need to route the 800 number that was dialed to the
right part inside of asterisk. I don't want to waste
a PSTN DID for each Watts number.
thanks
___
Asterisk-Users mailing list
I hope folks won't mind this.
We are looking for 3 serious programmer geeks.
Must beable to do:
Perl, C, PHP, programming
Can spell DNS and TCP/IP (Spelling BGP is a big plus)
You are self motivated, don't need to be constantly watch,
can work and play well with others, have a deep passion
If they are analog POTS type sets then any ATA (Sipura or Grandstream)
should work.
If they are a Digital set thats special for the PBX, then they
are a good door stop ;)
Basicly if you can plug it into a Tip/Ring analog line and it
works then it should with a ATA. Some sets use the yellow/blk
Hi List, incase anyone is interested, here are some specials
we are running. This page will get updated from time to time.
http://www.chagres.net/products/voip/specials.html
All instock and can ship today.
___
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[EMAIL
Hi List,
After reading a bunch of the docs, list post archives, it
still seems that a clear definition of how to clock the T100P
card is muddy.
zttool says that the link is INTERNALLY CLOCKED,
does this mean the T100P is providing clock, or does
this mean the T100P is getting clock from the T1
It appears that zttool doesn't actually report T1 span
errors.
If I inject BPV's, crc errors, framing errors, etc into
a T1 span, the counters on zttool don't change.
___
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[EMAIL PROTECTED]
THank you. Thats what I thought it should be.
Off to call the telco and tell them they are mucked up.
On Sun, Jan 11, 2004 at 06:54:11PM -0600, Don Pobanz wrote:
On Sunday, January 11, 2004 5:41 PM, John Brown (CV)
[SMTP:[EMAIL PROTECTED] wrote:
Hi List,
After reading a bunch
We sent $50 USD for the cause
john brown
chagres technologies
On Mon, Jan 12, 2004 at 12:10:09AM -0500, Andrew Thompson wrote:
Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, January 11, 2004 7:36 PM
Subject: [Asterisk-Users] More words for
For the list,
Mike received a partial order shipped 15-Dec, SN ending 4CD8.
Mike received email replies on 3-Dec and 17-Dec advising him
on his order.
Mike ack'd those emails.
This is the first time we have heard anything (phone calls or email)
from Mike since 17-Dec. Our CDR and SMTP logs
Hi List,
Matt hasn't contacted us directly about this. I've
responded to his previous statement that he hasn't
recevied the last 20 units, and never heard back from
him.
Matt, again, if this is an issue please do contact us.
Our CDR and SMTP logs show no such attempt.
Our inventory records
Technologies but I have read many many
posts to the same effect. If you are going to take someone's money then
follow through on your service or product in a timely manner. If you
cannot, close your business and stop taking people's money.
- Original Message -
From: John Brown
PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 3:17 PM
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
Just refund the guy his money...
- Original Message -
From: John Brown (CV) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 10
Hi List,
a number of our customers are reporting dropped calls.
here is the config.
1 T100P T1 Card
1 Asterisk (Mid Nov build)
T1 is signalled as a PRI(National)
The card will only sync up if we clock, if
we line side clock the card goes into yellow alarm
and won't sync up.
the only
...
- Original Message -
From: John Brown (CV) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 2:46 PM
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
For the list,
Mike received a partial order shipped 15-Dec, SN ending 4CD8
Your order was picked up on THursday by UPS.
All HT-286 orders have been filled.
On Fri, Dec 12, 2003 at 01:57:02AM -0500, Brian Capouch wrote:
John Brown (CV) wrote:
Hi List,
Just a quick note that we have cleared all back logs of Grandstream
product. If you have been awaiting
Hi list,
Well I really didn't want to see things get to this point,
but Sherman at Sipura along with their President Jan F.
leave me no other choice.
SIPURA has been provided a letter from our attorney for
Breach of Contract and damages. They have yet to respond.
A quick background.
1.
a very good and solid product.
cheers,
On Fri, Dec 12, 2003 at 11:20:58AM -0700, John Brown (CV) wrote:
Hi list,
Well I really didn't want to see things get to this point,
but Sherman at Sipura along with their President Jan F.
leave me no other choice.
SIPURA has been provided a letter
how companies are doing especially when they help the community.
On Wed, 2003-12-03 at 20:44, John Brown (CV) wrote:
Several things conspired to muck things up the last 3-4 weeks.
1. Surgery (repair of a previous hernia)
2. Travel to work at opening our EU warehouse
3. TSA dropping my
Several things conspired to muck things up the last 3-4 weeks.
1. Surgery (repair of a previous hernia)
2. Travel to work at opening our EU warehouse
3. TSA dropping my laptop, thus breaking my access to our VPN
4. New PRI going to a Asterisk box for our PBX and having the PRI be
mucked up.
]'.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Brown
(CV)
Sent: 15 November 2003 12:49
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] MWI and SNOM 200
Hi list,
how does one get a SNOM 200 MWI to work with * ??
When I press
The echo issues are line or PSTN.
make sure your tip and ring are correctly wired. Polarity
does matter and teh X100P does not do polarity fixing like
most consumer phones today.
john brown
chagres technologies, inc
http://www.chagres.net/products/voip/
On Sun, Nov 16, 2003 at 01:35:13PM
Technologies, Inc
Thanks,
Ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Brown
(CV)
Sent: Sunday, November 16, 2003 10:54 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
The echo issues are line
/RING and echo settings correct.
john brown
chagres technologies
On Sun, Nov 16, 2003 at 02:18:55PM -0500, Kevin wrote:
Just to be sure again, I did a reversal on the tip and ring with no
improvement.
-Original Message-
From: John Brown (CV) [mailto:[EMAIL PROTECTED]
Sent: Sunday
based on below, you have them in the same context
insert a context=foo line after channel =1
if you want channel 2 in a different context
On Sun, Nov 16, 2003 at 09:32:42PM +0200, Dan wrote:
Hi,
I have two X100P cards in the same system.
I can use both of them to initiate and/or
I replied privately back to Aaron. Seems our Spamassissin software
tagged his messages as spam. With the volume of spam email I've been
getting I haven't reviewed the spam folder in a bit.
I've noticed a couple of other emails got tagged as well and I'll
reply to those off list.
john brown
Hi List,
Here is the config
ext 2601 is a GS BT-101 phone
ext 2062 is a SNOM 200
latest public firmware on both
asterisk is Asterisk CVS-11/14/03-22:55:45
Make a call from 2601 - 2602 life good, call works
have 2602 place call on hold. The music on 2601 IS NOT
my music on hold. It
Hi list,
how does one get a SNOM 200 MWI to work with * ??
When I press the MWI button it doesn't connect with
voice mail on my * box.
thanks
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Any thoughts on how to make HOLD and TRANSFERs work
with a SNOM 200 and Asterisk ??
Thanks
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
So what do people think about adding the call rate to the CDR
structure??
This would allow you to detail a call with the rate that was
in affect for that call. When you come back later and do
the billing for the customer you would have the actual per min
rate in the record.
I think this solves
Billing Logs. 8-)
If your billing application can't handle this, it should be pretty easy
to build a small preprocessing application for the log files.
Why invent something new when you can reuse existing tools to do all or
most of the job?
On Sat, 2003-11-08 at 15:08, John Brown (CV) wrote
download the code,
complile the code,
start bashing on configs. :)
if you want to glue to the PSTN, i'd
recommend getting a FXO (WC-X100P) and FXS (TDM-10B)
card and some cheap SIP / VoIP phones (grandstream or snom)
john brown, ceo
chagres technologies, inc
Providers of VoIP hardware
anyone around here have the ability to terminate
a .NL phone number to IAX or SIP ??
if so please contact me off list.
thank you
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
so why does this work properly
[iax-ll]
type=friend
context=iax-ll-in
trunk=yes
host=iax2.ll.com
[iax-]
type=friend
auth=md5
secret=blahblah
context=iax--in
trunk=yes
host=rx8..net
and this doesn't
[iax-]
type=friend
auth=md5
secret=blahblah
context=iax--in
On Tue, Nov 04, 2003 at 05:48:38AM -0500, John Vozza wrote:
On Mon, 3 Nov 2003, John Brown (CV) wrote:
if you aren't running 1.0.3.81 or newer, then upgrade :)
Or NEWER Latest I can find is 1.0.3.81. Care to give us all an early
holiday gift? :)
newer stated for those that search
what version of GS firmware are you running ?
I call from PSTN to GS, GS does xfer to XTEN, hang up GS
call continues
if you aren't running 1.0.3.81 or newer, then upgrade :)
john brown
chagres
On Mon, Nov 03, 2003 at 06:15:02PM -0600, Steven Sokol wrote:
Hi,
Does anybody know how to
How does one send a broadcast message to all voice mail boxes?
I want to send a single message to every mailbox on the system
informing them of changes, etc.
any thoughts ??
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
press delete msg or delete thread and move on
On Sun, Oct 26, 2003 at 01:34:04PM -0300, CW_ASN wrote:
You see, guys? Some people loves to see masses of messy details and other
people don't... What can we do?
Regards,
Gus
- Original Message -
From: Bruce Ferrell [EMAIL
Hey, just thought the list might want to know that SIPURA
has released their really cool ATA device, the SPA-2000
If you are interested in purchasing these units we have
them cheaper than their own site does :)
surf over to http://www.chagres.net/products/voip/ata.html
We have also added
http == hyper text transport protocol
tftp == trivial FILE trasfer protocol
thus using tftp to do updates seems better. Its also
a smaller foot print code wise and in boot loader thats
important.
tftp servers are available,
On Wed, Oct 22, 2003 at 08:58:33AM +0100, WipeOut wrote:
John Brown
On Wed, Oct 22, 2003 at 02:24:57PM +0100, WipeOut wrote:
Here is another thought that I haven't heard mentioned...
How about changing the TFTP upgrade in favour of HTTP upgrades and
config file retrieval.. I am sure almost everyone has an HTTP server
available to them but I doubt many
On Wed, Oct 22, 2003 at 03:15:27PM +0100, WipeOut wrote:
http is a bad idea imho. I don't want to have to carry around
a web server on my laptop, or have to have my customers config
a web server to deal with updating their phone.
I would think setting up a web server would be easier than
Grandstream and Global IP Sound have inked a deal in which
Global IP Sound will provide its royalty free iLBC codec
to Grandstream. GS will integrate this codec into the
BT and HT product lines
___
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[EMAIL PROTECTED]
Can you provide more specific information. Saying Its Broke Jim
doesn't provide enough content :)
What version of code are you running on the GS ??
On Tue, Oct 21, 2003 at 01:33:48PM -0600, Steve Meyers wrote:
On Mon, 2003-10-20 at 20:38, John Brown (CV) wrote:
So please rate your ideas
Here is a quick tally of the various things people
asked for..
I'm going to go thru the list and weight the results
based on my scale of 1-10. This is just a count of
each item, otherwords how many times that item
came up. Some things I considered as bugs and lumped
them as bug-fixes
For the
Hi List,
I had a wonderful meeting with GS's President last week
and he is very interested in feedback on what top features,
functions, bugs the community would like to see in upcoming
firmware.
Please keep in mind that adding new features take time
to develop, test and such.
So please rate
If anyone buys GS phones from us (Chagres Technoloiges)
and runs into such problems, please let us know. We will
do what needs to be done to make it right.
I'll make sure that this feed back gets to Grandstream's
president..
John Brown, CEO
Chagres Technologies, Inc
On Mon, Oct 20, 2003 at
Hi Marcel,
Great news. Thanks for posting your success story
john brown
chagres technologies, inc
On Tue, Oct 14, 2003 at 01:02:17PM +0200, Marcel Prisi wrote:
Hi all,
Just a little note for the records and archives. We see many small
glitches / troubles in the mailing-list but rarely
Hi Josh, Costas, Adam, et all
We do sell the phones.
http://www.chagres.net/products/voip/phones.html
and digium cards
http://www.chagres.net/products/voip/cards.html
plus new things real soon :)
and if anyone ever has a problem, go yell at me
and I'll try like crazy to fix it.
john
Message --
From: John Brown (CV) [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Thu, 2 Oct 2003 09:06:00 -0600
Hi Josh, Costas, Adam, et all
We do sell the phones.
http://www.chagres.net/products/voip/phones.html
and digium cards
http
Provisioning System (GAPS). Free?
-- Original Message --
From: John Brown (CV) [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Thu, 2 Oct 2003 09:06:00 -0600
Hi Josh, Costas, Adam, et all
We do sell the phones.
http
--
From: John Brown (CV) [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Thu, 2 Oct 2003 09:06:00 -0600
Hi Josh, Costas, Adam, et all
We do sell the phones.
http://www.chagres.net/products/voip/phones.html
and digium cards
http
I'm looking for a way to manage large dial plans.
Blitz on IRC mentioned DynExtenDB
I'm wondering how stable it is since its not been
updated since 2002-12-15
Any other ideas ??
I want to have my dial plan in a SQL database
thanks
___
Yes, set the TFTP IP to 0.0.0.0
On Sat, Sep 27, 2003 at 11:10:50PM -0400, Uriel Carrasquilla wrote:
Is there anyway to prevent the BudgetTone from just doing a BIOS upgrade
without consulting?
I would be scared if my PC upgraded its BIOS at will.
Uriel
-Original Message-
From:
I'm not sure about ring tones
It does produce the 90 VRMS ring signal and
cause the analog phone to ring.
On Fri, Sep 26, 2003 at 09:55:45AM -0500, Peter Pauly wrote:
The PDF on the website says that this thing
supports a downloadable ring-tone. This
makes me somewhat suspicious - does
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