I installed one and works fine but of course when I try
to make the second call it says no lines are available
That's weird. I was under the impression the non-Digium ones didn't care how
many lines were in use, as there was no monitoring of such things in there.
Regards,
Chris
What caller id method is used in spain? Is it before or after the ring.
If you can set the ISDN termination box for UK caller id then the ID is sent
before the first ring.
on the sipura thats ETSI FSK with PR(UK)
Chris
- Original Message -
From: Manuel Dominguez [EMAIL PROTECTED
Have any providers started to offer T.38 yet? I am anxious to find a
solution for faxing.
--
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I have been using sipdiscount in sip mode (they are discontinuing their IAX2
connection) for a while. UK calls are free and its worked most of the time.
However, its not working this morning .-(
Chris
- Original Message -
From: RumaTech [EMAIL PROTECTED]
To: Asterisk Users Mailing
to a modtap used in structured
cabling environments.
Regards,
Chris
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Through. Try
setting the timing between 400 and 600 ms and also keep the Wait(1) in
the dialplan to give your PBX/KSU enough time to handle the flash.
Hope this helps you!
-Chris
--
Chris Shaw
IT Manager
Precision Pump, Inc
150 N Main St
Banks, OR 97106
Phone: 503-324-2361
Fax: 503-324-2203
I am venturing down this path soon, trying to
decide if I will be okay sticking with 2 x Fritz! PCI cards or if I should
really get a different card... 4 port ISDN or something...
Chan_Capi with 2 Fritz cards I'm told requires code
modifications to work etc...?
Suggestions?
Chris
to bite
the bullet and get a QuadBRI card from Junghanns. I'm inclined to give the
4xHFC option a try, simply because it's only an outlay of £60 or so compared
to a QuadBRI at over £300 last time I looked.
Regards,
Chris
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oping back to
the input. I suspect you have a bad phone, I have 100 of these phones
in use with asterisk and have no problems with them.
--
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NetConcepts
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This m
for
countries in Africa, and are there any companies out there selling them?
Thanks in advance.
Regards,
Chris
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JP Carballo wrote:
I would if I could. I'm paying more to Time Warner for a fraction of
the speed Verizon provides.
Until Verizon is done with their tedious wiring of affluent
neighborhoods, I can only dream.
Rich bastards. Come the revolution...
--
Chris Mason
NetConcepts
(264) 497-5670 Fax
Its ulaw.
sox -V foo.mp3 -t au -r 8000 -U -b -c 1 foo.ulaw resample -ql
is one way to get there.
You could also take a look at format_mp3 in asterisk-addons which is what I
use.
Chris
- Original Message -
From: Zach A [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non
to macro
outbound, passing it the number to dial, callerid to present, and any
number of routes in the order you want them to be tried. The macro dialout
just ensures that if the number called is genuinely busy, outbound doesn't
plough on with routes 2,3,4 regardless.
Hope that helps.
Regards,
Chris
a serious echo problem on a TDM
+ BT PSTN line in the UK
Thanks for your suggestions everyone
--
Chris Earle
System Solutions Specialist,
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Has upgrading to the newer Zaptel allowed you to use the newer improvements
in it? (sorry if that was implied)
Thanks for the speedy reply
Chris
- Original Message -
From: Adam Robins [EMAIL PROTECTED]
To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non
anything (DTMF tones too
low or whatever...)
Still wondering if it's an impedance issue...or something along those
lines/chipset..etc
I'm going to attempt upgrading Zaptel now, without upgrading the asterisk
Chris
- Original Message -
From: Chris Bagnall [EMAIL PROTECTED]
To: 'Chris
to have largely resolved the situation. Certainly my
experience has been that rxgain and txgain have far more impact on echo
reduction than any of the echo-specific settings. Get the gains right first,
then play with the echo-specific settings.
Regards,
Chris
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C.M. Bagnall, Director, Minotaur I.T
or Excel, depending on which supplier has
adequate stock at the time we need it.
In the UK, I'd highly recommend Rotronic as a good source of patch panels
and cable.
Regards,
Chris
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controller, so it seems this is
unrelated to Zaptel.
Regards,
Chris
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if Grandstream took this on board and provided an option for
call waiting, where, for example, the line 2 light flashed to indicate a
call, but there was no audible indicator.
Regards,
Chris
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calls (or however you do it) to include
SIP/211a as a member rather than 211.
Then use the phone's web config to assign a different ringtone to accounts 1
and 2.
It's a bit of a kludge, but it does work, and seems to be reliable.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
I am trying to construct a macro for long distance dialling. I have two
internet feeds, I have all routes including Teliax on Internet A and a
static route to Voxee on Internet B. I thought I could use the dialplan
entry below which uses the ChanIsAvail() command to check the
connection, but
patents aren't valid in your jurisdiction.
IANAL and you should take independent legal advice if you feel appropriate.
Regards,
Chris
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on a
frequent basis, the mac-based licencing method might prove cumbersome.
Regards,
Chris
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to enable the jitter buffer and have a lot of compensation.
I need a reliable provider I can use this with.
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?
Thanks for your suggestions everyone
Chris
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: Chris Earle (CBL) [EMAIL PROTECTED]
Sent: Thursday, January 19, 2006 1:02 PM
Subject: Re: [Asterisk-Users] Re: Connection TDM400P to UK PSTN
On Thu, Jan 19, 2006 at 11:30:25AM -0500
Under the regional tab (admin/advanced) select
ETSI FSK WITH PR (UK) for caller id method.
Make sure
PSTN CID For VoIP CID: is set to yes in the pstn tab to pass on the cid to
asterisk.
Chris
- Original Message -
From: Conrad Wood [EMAIL PROTECTED]
To: Asterisk Users Mailing
card to plug the phones in? These phones, isdn etc -- probably aren't
analog -- probably don't work with a TDM card right?
So I think what you were suggesting John is ISDN channel cards and a TDM in
the same machine? with * just bridging calls between the two?
Interesting. :-S
Chris Earle
Can the # be used as a valid key press for a user in a dial plan?
if so how can the asterisk recognize it as a valid key press?
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Echo Cancellers to MARK3.
Right, this is Asterisk 1.0.9, Zaptel 1.0.9.2 on Debian
Suggestions / Experiences in UK appreciated
--
Chris Earle
System Solutions Specialist,
- Original Message -
From: John Novack [EMAIL PROTECTED]
Newsgroups: gmane.comp.telephony.pbx.asterisk.user
Sent
box, Channelbank/telco-box, extension
handsets..
Thanks
--
Chris Earle
System Solutions Specialist,
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a number of similar entries
in my dialplans on 1.2.1 and they work perfectly.
Regards,
Chris
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I know there are numbers
provided by other providers in UK
If you happen to know of any, please feel free to post them. :-)
Regards,
Chris
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13, 2006 3:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] fax2mail
On Fri, 2006-01-13 at 15:03 -0600, Chris Stinson wrote:
Anyone here use fax2mail? Every fax get's flipped 90 degress. I was
just
wondering if anyone else had this issue and how
trying to do is the HT488 which does offer a true FXO
port. Alternative might be the Sipura SPA-3000, or of course, one of the
Digium TDM400 cards.
Regards,
Chris
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Anyone here use fax2mail? Every fax get's flipped 90 degress. I was just
wondering if anyone else had this issue and how they resolved it.
-
Chris Stinson
Network Operations Center
ISDN-Net, Inc.
615-221-4200 x103
[EMAIL PROTECTED]
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.
Any help appreciated.
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higher loads or higher conplexity
querries. It is good for flat file like problems. The
larger DBMSes are slower initially but scale better.
(Please CC any replies to my direct email)
Chris Albertson
Home: 310-376-1029 [EMAIL PROTECTED]
Cell: 310-990-7550
Office: 310-336-5189
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Chris Albertson
Home: 310-376-1029 [EMAIL PROTECTED]
Cell: 310-990-7550
Office: 310
will need to have the sound files in .ul, .gsm, or
whatever codec you use mostly. I only allow ulaw, so all of my MOH
files are .ul. This way it doesn't have to transcode at all.
That would seem to be the sensible answer.
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can cause this and what can I do to prevent it?
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The Austin Asterisk User Group is meeting next Monday. See
aaug.bybent.com for more information on the location. Topic and date
aren't correct on the page, but the location and time are.
--
Chris Tooley
512-646-1507
[EMAIL PROTECTED]
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Is there a TOC somewhere, so I can compare with the 1st version (which I
bought)?
http://www.osoft.com/store/productdetails.php?pid=151
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or
should the system be able to handle this loss of level?
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against minimal echo.
Will try that.
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it, and is there an alternative that is
stable?
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NetConcepts
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and you will be OK.
--- Kevin Bockman [EMAIL PROTECTED] wrote:
Chris Mason (Lists) wrote:
When I configured this server, I did not do the make mpg123 option.
Months later, I read about it and did it, as the client was asking
about
MOH. About a week later the server crashed, which it never has
,
Chris
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Messenger with any IM
clients apart from Trillian and Spark? (Trillian costs money and I'm not
that keen on Spark's lack of configurability)
Regards,
Chris
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we were discussing.
Where can I find the $300 SIP capable units?
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,
Chris
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minutes,
cross-network call charges skyrocket, whilst calls to the same network are
still competitively priced).
How might one go about configuring that in the dialplan?
Regards,
Chris
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Chris Bagnall wrote:
Has anyone using a GSM gateway incorporated some time monitors into their
dialplan?
For example, if a SIM card has 400 inclusive minutes to any network in a
month, I want to make sure that Asterisk doesn't go beyond that unless it's
for calls to the same network (once
had many SIP devices
behind a NAT (and by many I mean more than it's reasonably practical to
assign different port numbers to).
I'm no expert on firewalls, so hopefully someone'll correct me if I'm
mistaken.
Regards,
Chris
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This email is made from
, delivery is £17.37 for a 6-day delivery,
or £10 for a 30+ day delivery, both of which seem a bit high for an item
apparently located in the UK.
Regards,
Chris
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any QoS rules to both the SIP and RTP
streams.
Regards,
Chris
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trying to make phpagi stream a .gsm file from sounds dir. Asterisk is
answering the call and runnng the script. However no file to be heard
over the phone. Has anyone ever used the Asterisk phpagi and was there
any configurations that i should be aware of to stream the files? Oh all
to find any indication that this should not work the same on OS X as it
does on Linux. I'm wondering if anyone knows more about how STDIN/STDOUT
are used in Asterisk why the alarm timer might interrupt the read from
STDIN?
Thanks,
Chris
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) with the larger
font size.
Any thoughts / suggestions / pointers?
Thanks in advance.
Regards,
Chris
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approaches.
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That's right, it's a welltech. I have one working but when people call
in the ringing is not typical of American installations (indications?)
and it freaks people out. Also, I don't get callerid. Where can I get
the upgraded firmware?
--
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NetConcepts
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Hi,
I'm trying to get information on what the current status of TBCT support
in Asterisk is.
Thanks in advance,
Chris
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I've got one, they suck pretty bad.
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appreciated.
begin:vcard
fn:Chris Songer
n:Songer;Chris
org:Blaze Media Inc
email;internet:[EMAIL PROTECTED]
title:Database Administrator/ IT Director
tel;work:615-491-2459
tel;cell:615-491-2459
x-mozilla-html:TRUE
url:http://www.getblaze.com
version:2.1
end:vcard
=Enterprise_VoIPpname=VG400
Anyone had success with this?
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I am about to sent some Sipura 2002 ATAs out to a call center. I want to
use the dual line capability of the units, but I realize that the second
channel will not be able to use G729 simultaneously. What do you think
would the best option be for that channel?
--
Chris Mason
NetConcepts
(264
I have run my soundstation out of the soundpoint directory, and now get a
repeated error message of: Application sip.ld is not compatible with phone. I
found the warning AFTER turning the phone on within the network. Can this be
fixed?
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point that even comes close, at least this side of the
pond. I've tried the SPA-841, Aastra 9133i (more expensive, when you factor
PoE injector into costs), a Swisscom phone, etc.
Regards,
Chris
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.
Regards,
Chris
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address they provide in any way/shape/form.
I must point out I've no experience using the SIP side of the device at all.
I might try and steal one off a client in the new year and have a play with
it - I'd no idea it supported SIP at all.
Regards,
Chris
--
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will not be right
for the US, and they work out just as expensive as four SPA-3000s, which
are very good. Don't hold your breath for firmware updates.
With 4 X SPA3000, if one dies, you can replace just that one, and a
spare is a lot cheaper that a spare four port.
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e dropouts..
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To U
!
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Chris Albertson
Home: 310-376-1029 [EMAIL PROTECTED]
Cell: 310-990-7550
Office: 310-336
With this configuration I am getting choppy sound. What should they be
set to?
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Andrew Kohlsmith wrote:
Tell the Channel Banks clock to the line, and have the Sangoma card NOT sync
to anything (i.e. the A104 is the master, the channel banks the slaves).
I set the card up so that
Port1
TE_CLOCK= MASTER
TE_REF_CLOCK= 0
Port2
TE_CLOCK = NORMAL
TE_REF_CLOCK = 1
which
Ce fel de ajutor ai nevoie. Trimite-mi un
email.
Best regards,
Chris HARIGA
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of FaberK
Sent: Tuesday, December 20, 2005
11:07 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk
I have seen in several places where the analog adapters you can use for faxes
and modems have intermittent problems. Is this an issue with asterisk and are
there work-arounds?
We currently have side-stepped the issue by sharing the analog line between the
tdm card and the fax machine (I
/listinfo/asterisk-users
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512-646-1507
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that
an analogue phone would see?
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C F wrote:
I geuss this explains why you should test :)
It should work, you miscofigured something.
I am testing, it is a test setup. The ATA rings a phone connected to it,
and the Channel bank answers a call on an incoming CO line. Any ideas?
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Chris Mason
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reliably with asterisk.
Beyond that, I don't think the -486 really has a dialplan of its own in the
same way the Sipura units have, so I'm not really sure what else could be
causing the problem.
Regards,
Chris
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is changing but the phone
doesn't know, and it's religiously sending its refresh signals every x
seconds to the asterisk server.
The remote extensions are all defined with nat=yes in sip.conf
Any way to fix this (short of telling the clients they need a static IP) ?
Thanks in advance.
Regards,
Chris
through the
ATA, using an analogue phone.
The alarm panel does not seem to function properly thorugh the ATA,
either it is not going off-hook properly or the ATA is treating the
modem tones as a fax, I am not sure what is happening. Does anyone have
experience of getting this to work?
--
Chris
I know it's bad form to reply to one's own messages, but I should have added
that both boxes in question are running 1.2.
I was under the impression that many of the IAX jitter buffer issues had
been resolved in 1.2?
Regards,
Chris
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thoughts?
Regards,
Chris
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Jitterbuffer is to blame.
I feared as much. Thanks for the confirmation.
1) Is this a known issue with ILBC + Jitter Buffer?
2) Why does the same not occur from a SIP handset?
3) Is it safe or wise to disable the jitterbuffer?
Thanks kindly.
Regards,
Chris
--
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much heat.
P.S. - You should run AstLinux on your net4801:
http://www.astlinux.org
I very probably will. :-)
Regards,
Chris
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, shove a load of cordless phones on a
channel bank at convenient points around the site and contractors are never
far from a phone.
This is something we're hopefully doing for a property developer in the new
year. It'll be interesting to see how well it all works out.
Regards,
Chris
--
C.M. Bagnall
form factor I could see. I
have to wonder if a CF card based system can be adequate for this kind
of work, I have tended to move up to mirrored drives and a couple of
slots for that type of installation.
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NetConcepts
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send an email as it now does.
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http://lists.digium.com/mailman/listinfo/asterisk-users
HTTP or SMTP using a shell script, perl, etc
and/or via AGI (we use Bayham for MWI notifications to cellphones).
I don't beleive my country is listed, otherwise it would be very simple.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell
BRIstuff
installed (did it a few months ago to get n+201 branching on some dialplan
applications). Doesn't seem to cause a problem.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons
to answer.
Fortunately our requirement is only for 4-6 concurrent calls.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons
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Asterisk
anyone done anything like this? I also thought of setting
the CallerID Name to Server5 down for example.
Any ideas on this appreciated.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED
had placed a bid.
--
Chris Tooley
512-646-1507
[EMAIL PROTECTED]
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I've been using them for about 3 months and haven't experienced any
problems with them.
My only problem has been with my ISP. I get two calls going at the same
time and the ISP boggs down.
Regards,
Chris
- Original Message -
From: Rolf Brusletto [EMAIL PROTECTED]
To: Asterisk
/8/05, Chris Tooley [EMAIL PROTECTED] wrote:
gNumber has written an application, UnWired Buyer, based on
Asterisk. To
show our thanks, we would like to extend an offer to the
community. We
are currently offering an PayPal credit of $10 to everyone
are not required to win the auction but a bid must be placed
using the system.
For information on the program go to
http://www.unwiredbuyer.com/asterisk
--
Chris Tooley
512-646-1507
[EMAIL PROTECTED]
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you to learn how the system works, but there isn't a way to call in
and get the status of the auction.
Chris Tooley
On Thu, 2005-12-08 at 13:18 -0500, James Armstrong wrote:
Cool application, one question, can you call from any phone? Is the
number on the reference card an example, customer
automatically. :(
Chris Tooley
On Thu, 2005-12-08 at 13:43 -0500, Steve Totaro wrote:
Very neat, useful and extremely creative idea. I wish you much luck
with this product. One question I have and didn't see it documented.
What does the Right Price feature do? Also, can bids be scheduled
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