Anton Krall wrote:
Hi John, yes, Im using native transfer. What I do is use Monitor on the
dialplan of the extension that picks up the call coming from PSTN, so after
that, if the extension forward or transfers the call, monitor keeps
recording all thru the end of the call no matter where it
Anton Krall wrote:
Guys.
This idea has been banging my headfor days now and I feel the need to share
with you.
Imagine this scenario: all calls come in thru a receptionist, asterisk
records all incoming calls, the receptionist's work is to transfer the calls
to internal people but some
Hi;
I'm trying to record all inbound and outbound calls at a site, and I
have a problem with inbound calls that are transferred by a receptionist
using Snom's handset buttons (i.e. SIP transfer rather than using the
key sequences defined in features.conf).
The first leg of the call is recorded
FaberK wrote:
Hi guys,
maybe youìve got the answer...!
When a caller(not internal, but from PSTN) call *, I need to let him
hear a message, before * answer and the bill start running.
If is not clear, just let me know.
caller-telco(telco bill to the caller as soon as * answer)-asterisk
Andrew Kohlsmith wrote:
On Monday 20 March 2006 11:46, John Daragon wrote:
Alas, most (if not all) telcos object to you transmitting voice over
their circuits before they've started to charge you for the call.
Incorrect. I do this all the time with a PRI. You can't do this with POTS
Faris Raouf wrote:
Can anyone help point me in the right direction please?
I'm based in the UK and I want to start using a Premium Rate number with
Asterisk - I think the equivalent in the US would be a 900 number.
Effectively the caller pays much more to call such a number than a
normal
I'm getting a strange error on one of the two controllers on an AVM C2
card under chan_capi-cm-0.6.3.
I have two ISDN controllers defined, both in the same group, both
connections are UK ISDN2e Point to Point:
On the third outbound call (both of the first two calls are handled by
the second
Whoops ! sorry - wrong release ...
chan_capi-cm-0.6.4 !
John Daragon wrote:
I'm getting a strange error on one of the two controllers on an AVM C2
card under chan_capi-cm-0.6.3.
I have two ISDN controllers defined, both in the same group, both
connections are UK ISDN2e Point to Point
Armin Schindler wrote:
On Wed, 15 Mar 2006, John Daragon wrote:
Whoops ! sorry - wrong release ...
chan_capi-cm-0.6.4 !
There were problems with 0.6.3 and AVM cards, but should be fixed in 0.6.4.
Can you please create a full debug log (set verbose 5; capi debug) for such
a case
I've been asked to look at a tender for a switch, and one of the
capabilities the customer is looking for is support for Q.932. They
have a number of exchanges and are looking, in the future, to support
things like remote and aggregated operators.
Can anyone point me to an idiot's guide to Q.932
Marco Mouta wrote:
Hi all,
I've bought a TE110P, and received it today. So i decided to install
[EMAIL PROTECTED] 2.7 with this card.
In the past i had experiencies with X100P (clone card) and it never
take me so long to reboot the machine
Have you specified an inaccessible DNS
Phil Freed wrote:
I'm afraid that I am at a loss here. I am new to Asterisk, and have
successfully set up SIP. But I cannot get my FXS card working, and I'm
not sure what else I can try.
# modprobe wcfxo
/lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o: init_module: No such device
Hint:
Phil Freed wrote:
I have noticed a lot of folks mentioning the x100P, and very few
mentioning x100M (which is what I have). Are there important
differences between them?
The X100P was a PCI card with a single FXO port (actually a WinModem,
more or less).
The X100M is a daugterboard for the
Hi;
I've been asked to look at a large asterisk system implementation, which
would be a candidate for either a large cluster of PCs or a smaller
cluster based on Signate's SGI box(es).
I've waded through the requirements document, and I think I have more or
less all of the requirements covered
Sina, hi;
Let's just do a little recap.
You've downloaded zaptel-1.2.4 and done the
make linux26
make install
make config
thing on it. If you don't uncomment anything, the builds complete
without error and modules are installed in
/lib/modules/`uname -r`/extra.
You've performed the 2.6
higer speed modems, though...
jd
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(or your
alternative carrier) will have to enable it.
Anytime anyone asks for advice on a small system, someone wades in with
try [EMAIL PROTECTED]. I wouldn't recommend that myself.
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
Lambs Lawn
card.
Could someone confirm with me that this is the right direction to go -- ISDN
lines, Fritz cards/Asterisk box, Channelbank/telco-box, extension
handsets..
On the handset side you could use a couple of TDM4xx cards, or just use
SIP phones.
jd
--
John Daragon
Conrad Wood wrote:
Hi
I wonder whether anyone got the Sipura ata 3000 to decode British
Telecoms callerid and pass it to asterisk?
The userguide seems to suggest that this is not possible, is that right?
Current firmware handles it beautifully.
jd
--
John Daragon
hybrid PBX. We have an * system connected to two analog (FXS) ports via
a couple of Sipura SPA3000 ATAs, and we thought the Sipuras were failing
to detect call termination. Turns out that the default behaviour of an
FXS port on this PBX is *never* to hang up.
jd
--
John Daragon
connections (which isn't that difficult), you can build a
simple and scalable solution.
jd
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John Daragon [EMAIL PROTECTED]
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Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK
v
(for a similar sort of application), perhaps I'll do some
performance estimation up front and post it.
jd
-Original Message-
From: John Daragon [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 11, 2006 2:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk
/threads to handle the actual queries.
jd
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John Daragon [EMAIL PROTECTED]
argv[0] limited (Asterisk implementation consultancy)
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from a
locally coded application.
Responsive, easy to do business with, absolutely no problems at all.
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited (Asterisk implementation consultancy)
Lambs Lawn Cottage, Staple Fitzpaine
generated by BT rather than the ones the Grandstream generates.
With CAPI (which I tend to use), you add the b option in the dial
string. I presume bristuff does something similar.
exten = _0.,3,Dial(CAPI/g1/${EXTEN}/b) ; always ask for early B3
jd
--
John Daragon
Dogers wrote:
Quoting John Daragon [EMAIL PROTECTED]:
snip...
When you say ringtones, do you mean sounds like a UK phone when it
rings, or sounds like a UK phone when we ring someone else ?
It does actually sound okay when we ring someone else, but when it rings, it has
the long single
in theory, but I've never tried anything like this.
Is there any chance it would work?
Yep, we've done
ISDN2e --
Asterisk - H.323 - Cisco Call Manager
Analogue - Sipura SPA-3000 -
which worked really well.
jd
--
John Daragon
to
be specified as well. By default you get neither but both are
non-charegable upgrades (in our limited experience).
Our recent PTP installations have turned up without CLIP but
with COLP. YMMV, it seems...
jd
--
John Daragon [EMAIL PROTECTED]
argv
).
I'd second that. For a single ISDN2e connection the AVM Fritz card is
really hard to beat/
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
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to explain this; does anyone understand
why it might be ?
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
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Tony Mountifield wrote:
In article [EMAIL PROTECTED], John Daragon [EMAIL PROTECTED] wrote:
Hi;
I've been looking for an arbitrary way of discovering when the last
user has left a Meetme conference...
It occurred to me that I could launch an agi script to keep watch over
the conference
from
that it's fine. Range is about 100m in clear air. We have 3 ft thick
stone walls and it copes with that very well.
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
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much in terms of interrupt requirement, CPU load c ?
Any info gratefully received.
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
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function called from invalid context at
include/asm/semaphore.h:107
in_atomic():1[expected: 0], irqs_disabled():0
..
..
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
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(taking, IIRC, the settings in the page and actually applying them to
the hardware...)
This seemed to fix all of the echo issues we'd seen.
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
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v
x 64 bit PCI-X and a 32 bit PCI slot. It doesn't
seem to matter what we disable in BIOS, both PCI slots share an
interrupt. We can chose *which* interrupt they share, but they always
share one. You haven't found a way around this (BIOS upgrade/whatever)
have you ?
jd
--
John Daragon
channel) GBP 182.32 DDI Non Quota
GBP 208.32 DDI Quota
jd
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John Daragon [EMAIL PROTECTED]
argv[0] limited
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before the CID has been
detected. But you've obviously thought of that - hence the delay...
It may be worth firing up ethereal to check that the CID really isn't in
the INVITE.
Are you using version 3.1.7 of the Sipura firmware?
jd
--
John Daragon
.
jd
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John Daragon [EMAIL PROTECTED]
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Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK
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with the current firmware.
Have you got PSTN CID for VOIP CID set to yes ?
jd
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John Daragon [EMAIL PROTECTED]
argv[0] limited
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Kerry Garrison wrote:
Yes I do have that set to Yes.
Does the SPA-3000 show the caller ID in the last call field in the
summary page ? It's capable of interpreting a bewildering array of
callerid schemes - is it set to what your local telco is generating ?
jd
--
John Daragon
, I
think, the protection circuitry.
Here's the question, then : what (apart from jumping through regulatory
hoops) is to stop a simple array of MOSFETS (and a bit of control
circuitry) implementing a failover switch controlled (say) by a pin on a
serial or parallel port ?
jd
--
John
Sergio Serrano wrote:
http://www.junghanns.net/en/ISDNguard_produkt.html
srsergio
-Mensaje original-
De: John Daragon [mailto:[EMAIL PROTECTED]
Enviado el: jueves, 20 de octubre de 2005 17:24
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] E1
feeling that all the
reframing stuff just adds another set of variables if something goes
wrong somewhere.
Thanks again.
jd
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John Daragon [EMAIL PROTECTED]
argv[0] limited
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softdtmf=0
accountcode=
context=from-pstn
echosquelch=0
echocancel=yes
echotail=64
devices=2
Hope this helps...
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK
v +44 (0) 1460 234068 f +44
= 0x3302
Oct 6 20:04:23 VERBOSE[12454]: == DISCONNECT_IND PLCI=0x202 REASON=0x3302
Oct 6 20:04:23 VERBOSE[12454]: sent DISCONNECT_RESP PLCI=0x202
Oct 6 20:04:23 VERBOSE[12454]: -- CAPI Hangingup
TIA
jd
--
John Daragon
an Eicon DIVA card running under AAH ? Or
have any idea why this is happening ?
TIA.
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
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Armin Schindler wrote:
On Sun, 2 Oct 2005, John Daragon wrote:
Hi;
I've got an AAH installation where a customer wants to install an active Eicon
DIVA BRI card. AAH is built on Centos 3.5 which is currently at kernel
2.4.21.37. Support for Eicon active cards is built-in.
I've debugged
with the
SPA-3000s, but you would, of course, lose a lot of flexibility...
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
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describing is 2 x ISDN30 connections (around 2Mbit/s
each ?)
I'm not familiar with the SA telephone system, but in the rest of the
world (more or less) the card you'd need is the Digium TE110P which is
switchable between T1 (24 channel) and E1 (30 channel) ISDN.
jd
--
John Daragon
for Debian or SuSe.
What is the general concensus on the best Linux to run
Asterisk with CAPI?
SUSE (as far as I know) is the only distro that really *expects* you to
be using ISDN2e as a matter of course.
jd
--
John Daragon [EMAIL PROTECTED]
argv[0
Tzafrir Cohen wrote:
On Wed, Sep 07, 2005 at 10:10:05AM +0100, John Daragon wrote:
YT Lim wrote:
We have tried Asterisk 1.0.9 on FC4 and have never
been able to get CAPI (with Fritz card, fcpci) to work
properly. Apart from that Asterisk works fine in
switching internal calls. But it's
://lists.digium.com/pipermail/asterisk-users/2005-March/096754.html
jd
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John Daragon [EMAIL PROTECTED]
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trunk because of the sharing of RTP
overhead ?
jd
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John Daragon [EMAIL PROTECTED]
argv[0] limited
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Chris Mason (Lists) wrote:
John Daragon wrote:
Hi;
I'm looking for a Polycom distributor in the UK who can supply a small
number (around 20) IP301 / IP501 handsets. Can anyone recommend someone ?
jd
I have been buying from Zycko - very efficient and on the ball.
Ta.
jd
--
John
out which network a number is on, so this may not work
for you ...) might it not be cheaper to get hold of a 3G gateway and
route at least some of your calls directly over the relevant mblie network ?
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
Peter Valkov wrote:
John Daragon wrote:
Peter Valkov wrote:
I have build asterisk from latest CVS HEAD-05/09/05 with H323 support
as described in README file.
Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2
kernel-2.6.11
I tested it with following phones: -- XLite
ring
Peter, hi;
I haven't looked at the openh323 code, and I might not get time to...
but in my limited experience, 60 second delays are almost always DNS
timeouts.
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
Lambs Lawn Cottage, Staple Fitzpaine
John Daragon wrote:
Peter Valkov wrote:
I have build asterisk from latest CVS HEAD-05/09/05 with H323 support
as described in README file.
Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2
kernel-2.6.11
I tested it with following phones: -- XLite (SIP softphone)
-- QMix SIP IP
, but problematic...
jd
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John Daragon [EMAIL PROTECTED]
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___
Asterisk
on my time!).
jd
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John Daragon [EMAIL PROTECTED]
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___
Asterisk
RedHat 8.0,
but I'm having difficulty finding information on running it under the Fedora
2.6 kernel. Is there anyone out there running this combination ?
jd
--
John Daragon argv[0] limited [EMAIL PROTECTED]
Lambs Lawn Cottage, Staple Fitzpaine, Taunton TA3 5SL, UK
(house
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