-channel of span 1
My zaptel.conf file:
span=1,0,0,ccs,hdb3,crc4
Try: span=1,1,0,ccs,hdb3,crc4
That tells the card to take its signal timing from the remote end.
Cheers
Tony
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(which was s), but in the current context (i.e. [whatever-its-in-already]).
The term stack doesn't imply any kind of subroutine call-and-return
functionality.
Hope this helps
Cheers
Tony
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,
but generate MOH locally.
My BTs do SIP hold, and the caller receives Asterisk MoH. It's possible
that earlier firmwares didn't do it, but I didn't try it till I was
running 1.0.5.9, and I'm pretty sure it worked then.
Cheers
Tony
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the zaptel.conf file is set up correctly for
the combination of FXS/FXO modules you have, and that the
driver is successfully loaded.
Cheers
Tony
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I have a potential client that wants to send many faxes simultaneously,
over E1 trunks.
How CPU intensive is spandsp's txfax? How many concurrent faxes could
be sent by a decent CPU (e.g. Xeon 3GHz) before timing starts to get
disrupted?
Cheers
Tony
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the card is idle.
Is this normal behaviour, or does it signify a problem?
Normal behaviour. Once an hour, all idle channels get restarted, but
active channels are left undisturbed.
Cheers
Tony
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you have BINARY mode enabled.
Seshu
NOTICE: If received in error, please destroy and notify sender.
But why destroy the sender as well as notifying them?
Cheers
Tony
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In article [EMAIL PROTECTED],
Kenneth Porter [EMAIL PROTECTED] wrote:
--On Thursday, March 31, 2005 4:52 PM + Tony Mountifield
[EMAIL PROTECTED] wrote:
Try downloading again. If using FTP, ensure you have BINARY mode enabled.
I question that. A corrupted download would lead to all
use trn to read and post too, as I have yet to find anything that
is as fast to use.
Cheers
Tony
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that if there was a genuine preference amongst the
majority for a web forum instead of a mailing list, there is more than
enough skill and resources to make it happen. The fact that it hasn't
happened might just say something.
Cheers
Tony
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In article [EMAIL PROTECTED],
Tim Bass [EMAIL PROTECTED] wrote:
I just joined this list yesterday,
And already you are telling the rest of us we're doing it all wrong.
Great.
Cheers
Tony
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for me. Perhaps there is a problem with your
system or your ISP's mail servers. Or maybe the list server knows you
don't like mailing lists and is just doing it to spite you :-)
Cheers
Tony
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In article [EMAIL PROTECTED],
Bruno Hertz [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] (Tony Mountifield) writes:
I totally agree. I run a local INN server and all the mailing lists I
subscribe to get turned locally into newsgroup postings in moderated
groups. When I make a posting, it gets
In article [EMAIL PROTECTED],
Bruno Hertz [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] (Tony Mountifield) writes:
Yes, based on a standard install of the INN rpm in Red Hat or Fedora.
I've just put together a page with a description and links to the two
perl scripts used. See http
In article [EMAIL PROTECTED],
Tim Bass [EMAIL PROTECTED] wrote:
Tom Ivar Helbekkmo. Grow up and stop posting to this tread.
pot calling the kettle black
Cheers
Tony
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command with a NoOp command as follows:
NoOp(DIALSTATUS=${DIALSTATUS})
Then it shows up in the log. It also provides something for my Manager event
parser to see, in order to discover the reason for a failed call.
Cheers
Tony
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Play
the heading
FreeTDS stuff.
If you do need it, then you will probably have to get a newer FreeTDS.
Cheers
Tony
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restrict the range of codecs that will
be considered for the call?
Thanks
Tony
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on, but I've certainly
had no issues with either card (Sangoma A101u vs T100P and TE405P).
Do the Sangoma cards use zaptel-compatible drivers or something different?
Do they provide a timing source in the same way as Digium cards do?
Cheers
Tony
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in a 3.3V
slot and still working?
I have no intention of hacksawing a board myself, but the findings of a
year ago suggested that all Digium would need to do is to respin the PCB
with an extra slot and make no other changes.
Cheers
Tony
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In article [EMAIL PROTECTED],
Matt Ryanczak [EMAIL PROTECTED] wrote:
meetme work in gsm only,
Sorry, that's completely wrong. There is no gsm in meetme.
Meetme works partly in uLaw and partly in Signed Linear (but I have
never worked out exactly which parts are in which)
Cheers
Tony
--
Tony
Andrew, thanks for your comments...
In article [EMAIL PROTECTED],
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On April 7, 2005 04:02 pm, Tony Mountifield wrote:
That's a pity and I'm not convinced the assertion is true.
Andrew, if you read this, is your hacksawed TE405P board still in a 3.3V
is that it is drift due to
clock differences between the ZAP timer and the SIP device, but am
not sure how to determine when a frame should be dropped or duplicated.
Cheers
Tony
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why no RTP is being generated?
Any suggestions would be appreciated!
Cheers
Tony
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,Macro(outisbusy); No available circuits
exten = _NXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten = _NXX,2,Macro(outisbusy); No available circuits
Couldn't you have just put the w in once, in the Dial command that
is inside [macro-dialout-trunk] ?
Cheers
Tony
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/include/tds.h ]; then echo cdr_tds.so; fi)
MODS+=$(shell if [ -f /usr/local/include/tds.h ]; then echo cdr_tds.so; fi)
Cheers
Tony
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have, perhaps you could post the contents of [macro-dialout-trunk].
Cheers
Tony
Robert Andrew Keller
Ferndale School District #502
[EMAIL PROTECTED]
360-383-9228 PH.
360-383-9218 FAX
Paving the way for tomorrows genius.
From: [EMAIL PROTECTED] (Tony Mountifield)
Organization: Software
-minutes) in new stack
-- Playing 'this-conf-will-end-in-5-minutes' (language 'en')
== Manager 'MeetMe' logged off from 127.0.0.1
So it appears that my variable ${confNo} is not being set, or at least
honored.
Any thoughts?
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, as I have a feeling it will be much less hungry for file
descriptors!
Comments, anyone?
Cheers
Tony
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In article [EMAIL PROTECTED],
Michael Manousos [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
When testing the ability of dual 3GHz Xeons to handle many simultaneous
OH323 calls (G.711 so no heavy transcoding), I discovered that chan_oh323
is EXTREMELY profligate with file descriptors
In article [EMAIL PROTECTED],
Henry Jensen [EMAIL PROTECTED] wrote:
Is there any way I can switch the TE110P card to NT-Mode ?
In /etc/asterisk/zapata.conf, change signalling=pri_cpe
to signalling=pri_net
Cheers
Tony
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Play
In article [EMAIL PROTECTED],
Henry Jensen [EMAIL PROTECTED] wrote:
On Tue, Apr 12, 2005 at 12:23:52PM +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Henry Jensen [EMAIL PROTECTED] wrote:
Is there any way I can switch the TE110P card to NT-Mode ?
In /etc/asterisk
and userInputMode settings in
the oh323.conf file.
Cheers
Tony
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, to no avail.
So what's going on?
We can't say for certain, because you've only provided an approximation
of what you're running, rather than actually what you're running.
But I hope the suggestions above do help.
Cheers
Tony
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. The remote party hung up.
Neither of those scenarios requires you to call Hangup again.
If you want to ensure you have an h extension, but don't need it to
do anything else, just use:
exten = h,1,NoOp
Cheers
Tony
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,Hangup cured the problem.
This was several months ago; I haven't tried reproducing it in more recent
versions.
Cheers
Tony
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In article [EMAIL PROTECTED],
[EMAIL PROTECTED] wrote:
On Wed, 13 Apr 2005, Tony Mountifield wrote:
If you want to ensure you have an h extension, but don't need it to
do anything else, just use:
exten = h,1,NoOp
but watch out for your CDR records. In my experience, h
in another computer, e.g. an Ethernet card. See if it
initializes and generates interrupts. If not, then your problem is
on the PCI bus somewhere.
Cheers
Tony
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In article [EMAIL PROTECTED],
Paul Brock [EMAIL PROTECTED] wrote:
Finally, Anyone know of a Digium hardware Reseller in the Uk at all??
www.telappliant.com
Cheers
Tony
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)
revision 1.1
date: 2004/08/01 14:19:04; author: markster; state: Exp;
Rename newp to newpvt (bug #2190), change hold music.
=
Try checking out (or updating) with just v1-0 instead.
Cheers,
Tony
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demonstrated that a standard LAN card was
also unable to generate an interrupt, the vendor accepted that it could
be a problem with their backplane, and came up with a hardware fix.
Cheers
Tony
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, or else
the copy of Asterisk must be commercially licensed.
Code that is a separate executable, and only communicates with Asterisk
via a communication channel (e.g. AGI, Manager API, etc) does not have
to be GPL in order to be used with a GPL Asterisk.
Cheers
Tony
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channel and into a slave conference
(where the participants just listen to it).
Any advice on good ways to approach this would be much appreciated!
Cheers
Tony
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that just waits to get DTMF for up to 10
; seconds due to the ResponseTimeout
exten = t,1,Goto(somewhere-due-to-timeout)
What's the reason for having a zero-length Wait befor the Answer?
Cheers
Tony
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as follows, and putting it into /var/spool/asterisk/outgoing:
Channel: Local/[EMAIL PROTECTED]
Context: phones
Extension: 2100
Priority: 1
That will ring extension 2000, and then when it answers it will ring
extension 2100 and connect them together.
Hope this helps
Cheers
Tony
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be easy to find as it's so recent.
Cheers
Tony
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manager method instead??
Yes, send the line Action: Originate, then send the same data as in
the .call file, and follow it with a blank line to terminate the request.
Make sure you end lines with \r\n, not just \n
Cheers
Tony
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. From then onwards, Firefly cannot receive calls, only place
them.
Cheers
Tony
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in the behaviour described. Without knowing your particular setup, it
is impossible to know what the cause could be. Perhaps you could
describe in more detail.
Cheers
Tony
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to miss one!
Cheers
Tony
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for failed
calls (busy, etc) comes in on the wrong channel. I have to work out a way
of tracking those over the next day or two!
Cheers
Tony
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My colleague was told today that (some?) Cisco VoIP kit supports IAX.
I found that hard to believe. Was it likely the talk of an over-eager
salesman, or is there some truth in it?
Cheers
Tony
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,resp_created)
VALUES(${advertid},\'${CALLERIDNUM}\',\'${RECORDED_FILE}\',NOW())')
exten = h,3,MYSQL(Disconnect ${conn})
Hope this helps!
Cheers
Tony
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? kernel IPtables? Also, where
is it appearing? e.g. /var/log/messages, console, or Asterisk log file.
What does your iax.conf look like?
Cheers
Tony
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Mark Benson [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
Mark Benson [EMAIL PROTECTED] wrote:
Have a problem that I have been battling with for a few days now with
help from voiptalk.org support.but I thought someone here might have
seen this before.
I have an asterisk box running
.
In the case of voiptalk, friend isn't useful unless you name your outgoing
section [08700nn] and use Dial(IAX2/08700nn/${EXTEN}). In that
case you could use type=friend, but I think that makes things less clear.
Hope this helps!
Cheers
Tony
Cheers,
Mark
Tony Mountifield wrote:
Mark
In article [EMAIL PROTECTED],
Peter Childs [EMAIL PROTECTED] wrote:
Contact Digium Support. They have been very helpful with this issue
(mention your using the G4 server with the Intel E7520 Chipset..)
So do they have a solution? What is it?
Cheers
Tony
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obvious?
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RTP stream to clock the
outgoing. Are you saying that Asterisk can now handle silence suppression
on RTP streams if a Zaptel timing source is available?
Cheers
Tony
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Hi,
Is anyone here using the SuperMicro SuperServer 6014P-8R with Asterisk?
I'm especially interested if you've used it with a TE405P or TE410P.
Cheers
Tony
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,
a local MySQL server and a remote MS SQL Server (using cdr_tds).
I have just verified that a recent call appears in all three places.
Cheers
Tony
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cdr in two databases
Tony Mountifield wrote:
I beg to differ. My test system here logs CDRs simultaneously to
Master.csv,
a local MySQL server and a remote MS SQL Server (using cdr_tds).
I'm glad to be corrected. Thanks!
___
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, while new features will again be added to CVS HEAD (1.3?).
Someone please correct me if I'm wrong.
Cheers
Tony
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.jpg
for an example.
If I do asterisk -rv on a normal login, either via the console
or an xterm, the text appears correctly.
Does anyone have any ideas what is causing this and how to fix it?
Thanks
Tony
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In article [EMAIL PROTECTED],
Kristian Kielhofner [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
I've just set up a new box with FC1+updates and the latest Stable
Asterisk from CVS.
Why are you using FC1 when FC3 is out? Better yet, why are you using
FCx at all?
Thanks for your
In article [EMAIL PROTECTED],
Carlos Chavez [EMAIL PROTECTED] wrote:
On Mon, 28 Feb 2005 20:58:48 + (UTC), Tony Mountifield wrote
I've just set up a new box with FC1+updates and the latest Stable
Asterisk from CVS.
Asterisk is started with the default safe_asterisk script
In article [EMAIL PROTECTED],
Kristian Kielhofner [EMAIL PROTECTED] wrote:
Howard Lowndes wrote:
On Tue, 2005-03-01 at 08:03, Kristian Kielhofner wrote:
Tony Mountifield wrote:
I've just set up a new box with FC1+updates and the latest Stable
Asterisk from CVS.
Why are you using
or is necessary to
make variable that is somehow unique to each call ???
http://www.voip-info.org/wiki-Asterisk+cmd+SetVar tells you the answer is that
each call gets its own variable space.
There is also a global variable space which you can write to using SetGlobalVar.
Cheers
Tony
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In article [EMAIL PROTECTED],
Nigel Taylor [EMAIL PROTECTED] wrote:
Can anyone recommend a Digium Reseller in the UK ?
TelAppliant
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of ways to tackle the problem, and to
determine whether it really is the Asterisk bridges or the phone systems,
I would be very, very grateful, as it is turning into a nightmare!
Cheers
Tony
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Can anyone advise from experience what size of PC would be needed
to support two TE405P 4xE1 cards to provide conference bridging
for up to 20 concurrent conferences of 10 participants each?
All the participants would be on the E1 trunks, not VoIP.
Thanks in advance,
Tony
--
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haven't found MoH to give any trouble. Do you have a Zaptel card
in your system? Music on Hold needs a timer, which is normally provided
by the zaptel driver. If not, you will need to use ztdummy or zaprtc.
See http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer
Cheers,
Tony
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with your problem though.
Cheers
Tony
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fine).
Can anyone suggest what things I should check or change?
Cheers
Tony
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to send DTMF via via RTP (RFC2833) with a payload
type of 101. (tried 100 and 102)
The first 2 codecs are set to PCMU and PCMA (tried to switch those
arround too).
Put dtmfmode=info in your sip.conf, and set the phone to use SIP INFO.
Then it will work.
Cheers,
Tony
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Work
of package X, you need to have package X-devel installed.
Cheers
Tony
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kernel modules (zaptel, zaprtc)
should also be built on FC1 using gcc32, not just cc or gcc. This
also gets rid of all those type punning warnings.
Cheers,
Tony
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is installed. It's only kernel modules
that need to be compiled with gcc32, but there's no harm in trying it
for applications too, as, far as I know.
Cheers,
Tony
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Tony Mountifield
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In article [EMAIL PROTECTED],
Leo Ann Boon [EMAIL PROTECTED] wrote:
The new TDM400P with FXO doesn't take up any IRQ. I've 2 boards and both
are not using any IRQ.
Weird - does that mean they can't provide Zaptel timing like the X100P can?
Cheers
Tony
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Tony Mountifield
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, work fine , just
chunk error message
checksum before = db8e
checksum after = 4db2
checksum failed
Are you running the perl program on Unix/Linux or on Windows?
It has only been tested on Linux, and may need binmode STDIN;
if running under Windows.
Cheers,
Tony
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with the extra 3 bytes, and another with the extra 1 byte?
There may be something my program hasn't taken account of, and it would
help me to find out what it is.
Thanks
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
store.yahoo.com or www.digium.com
you will be helping to support Digium, who gave Asterisk PBX to the
community. This is a Good Thing (TM).
The other three are OEM copies of the same card, but do not benefit
Digium at all.
Cheers,
Tony
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Tony Mountifield
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sure makering.pl has the x bit turned on, wherever
the file is located.
Cheers
Tony
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the PostgreSQL sections,
and it looked like you also didn't have zaptel installed.
Cheers
Tony
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must be generated as ring1.bin
and not just renamed from another file name. I haven't tested this
theory.
You then need to go to the phone's web page to tell it to use the new
ringtone.
Cheers
Tony
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Tony Mountifield
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somewhere visible?
Cheers
Tony
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Tony Mountifield
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messy
Cheers
Tony
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don't you want to use sox? I see from http://sox.sourceforge.net/
that it is available for Windows, and I would expect that as it compiles
for BSD it would also compile for Mac OSX.
Cheers
Tony
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Tony Mountifield
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HT286, and it seems to behave much
better.
I have two BT102 phones running 1.0.4.68, and one of them still does it.
Register Expiration is set to 3 minutes.
Cheers
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
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and it sounds like one or two 3GHz CPUs should do it, but if anyone
reading has direct experience of this kind of application, I'd be
very grateful for any comments.
Thanks
Tony
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a make in asterisk-addons.
Cheers
Tony
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still
make outgoing calls.
Cheers
Tony
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compiles.
Cheers
Tony
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through the last few
weeks' messages to see if your question has already been answered
recently.
See http://lists.digium.com/pipermail/asterisk-users/
Cheers
Tony
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by the
originator to place a call via the trunk is the same as that used by the
trunk to deliver a call to the destination. Is that correct?
There's a little more information about this on the Wiki:
http://www.voip-info.org/wiki-Asterisk+dimensioning
Thanks, I'll have another read.
Cheers
Tony
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Tony
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