section if you get a *.
Cheers
Tony
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
Hi all
I'm just having a problem now and I don't have any idea how to do this.
It is pretty simple. When a customer calls, to speed up the navigation
}.
However, for compatibility, the old jump behaviour can be reinstated with
the j option.
The above applies when Dial is called from the dialplan. I have no idea
whether it applied when Dial was called from EXEC in AGI, but it's worth
a try.
Cheers
Tony
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, then I would go back to Asterisk
1.4 instead of using an old beta of 1.6.0. Many things have been
fixed since 1.6.0-beta9.
If I'm installing a new system based on the latest Asterisk 1.4, should
I use zaptel or dahdi with it? Which version?
Cheers
Tony
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, and Zaptel 1.4 with Asterisk 1.4.
Cheers
Tony
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I am subscribed to several of the mailing lists hosted at lists.digium.com.
However, the memberships reminder that I receive on the first of each month
only lists asterisk-biz, and none of the others.
Just curious whether this was intentional or a mis-configuration.
Cheers
Tony
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In article [EMAIL PROTECTED],
Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Sep 01, 2008 at 10:43:51AM +, Tony Mountifield wrote:
I am subscribed to several of the mailing lists hosted at lists.digium.com.
However, the memberships reminder that I receive on the first of each month
only
In article [EMAIL PROTECTED],
Kevin P. Fleming [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
Interesting... I would have expected a single email from the site, listing
all the groups I am subscribed to. I'm sure I've had that from other
mailman-driven list sites in the past, when I've
reminded plenty and I don't really need the additional noise.
You mean like ONE extra message a month? For 50 messages a day, that
is approximately a 0.066% increase!
Cheers
Tony
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tried later versions.
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Tony
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do not know to whom belongs the ANSWEREDTIME
value :
exten = h,1,DeadAGI(myagi.agi,0,${DIALEDTIME},${ANSWEREDTIME},00)
Use the value of ${UNIQUEID} to distinguish between the channels. It will
be unique for every channel in a system.
Cheers
Tony
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In article [EMAIL PROTECTED], Drew Gibson [EMAIL PROTECTED] wrote:
Kevin P. Fleming wrote:
Tony Mountifield wrote:
My guess is that 1.2.24 will work, which was at revision 3842 of the tree
(rev 3741 of wctdm.c). The next change to wctdm.c (rev 4126) looks
innocuous
enough
:-)
Cheers
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in the 1.2 series that
affects ring detection on the TDM400P FXO ports? Any critical settings
in zaptel.conf or zapata.conf?
For now, I've had to revert to the original versions, in which ringing
once again works fine.
Cheers
Tony
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In article [EMAIL PROTECTED], Drew Gibson [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
A customer has an Asterisk box with two TDM400P cards, running [EMAIL
PROTECTED] 2.8,
which has Asterisk 1.2.7.1 and Zaptel 1.2.5. This has been running ok
for a while, with only some small issues
(or whatever the IP of your remote H323 endpoint is)
port=1720
If that still doesn't help, please mention what versions of asterisk and
asterisk-addons you are using.
Cheers
Tony
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to it, but I don't think such an
application has made it into official versions yet (1.4 or trunk).
Cheers
Tony
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timeslot?
Not sure where you got the idea to use 1-2 as the channel number.
Just use Zap/1/517255333 to call on channel 1, Zap/2/517255333 for
channel 2, etc.
The channels are as numbered in /etc/zaptel.conf and /etc/asterisk/zapata.conf
Cheers
Tony
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whether Objective Systems have abandoned
chan_ooh323 and
the ooh323c stack, but it would be great to see them
moved from -addons
into the main Asterisk tree.
Cheers
Tony
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the verbose output from the log file.
Cheers
Tony
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chan_ooh323 and
the ooh323c stack, but it would be great to see them moved from -addons
into the main Asterisk tree.
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Tony
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two Asterisk boxes, so am
just after some confidence regarding the Avaya, or any gotchas to be
aware of.
Cheers
Tony
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different things for each.
Cheers
Tony
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got to about 6 or 7 calls through it.
Eventually I discovered echo cancellation was turned on, causing the
1ms interrupt service routines to take more than 1ms to execute!
It's much happier with no echo cancellation, although I haven't tried
running all 120 channels at once!
Cheers
Tony
--
Tony
in dial cmd only one party could
hear?
Another interesting issue is, if I use Answer() command at the begining the
sound becomes one way even if I use t in options.
Try adding reinvite=no to the sip.conf or users.conf definition for your
SIP service provider.
Cheers
Tony
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(but best to do it
for the Service Provider endpoint) tells Asterisk never to optimise
itself out of the media path, even if t and T are not specified.
Cheers
Tony
Cheers,
Moe
On Sun, May 18, 2008 at 1:14 AM, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED
cards? Or are wanpipe and wanrouter generic
software components that both companies just happen to use?
Cheers
Tony
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with colors!
Cheers
Tony
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time this week, and am at
last beginning to understand why it does things the way it does. :-)
It's a good source of ideas as well as something to use in its own right.
Cheers
Tony
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to hi-Z if anyone knows.
Cheers
Tony
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Does anyone know if the Digium PRI cards can be configured or modified
to have a high-impedance input on the RX pair? I would be interested in
this in order to build a bi-directional PRI audio sniffer using two
E1/T1 ports per trunk to be monitored.
Cheers
Tony
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of National.
The second set of parameters sends all numbers as-is, but with a TON of
Unknown. This may allow the exchange to do normal interpretation like it
does with analogue phones, i.e. 00CC for international and 0[1-9]...
for national.
Cheers
Tony
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to as VAD, bandwidth saving, or something similar).
2. If ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable
the line internal_timing=yes. That should make it play out based on
internal zaptel timing instead of timing off the incoming stream, I think.
Cheers
Tony
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. This is with ztdummy running.
What else is going on on this server? Does it have any virtual machines
on it? Does it have X Windows running? What does top show?
Cheers
Tony
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is that the kernel must have been built with
CONFIG_RTC and not CONFIG_GENRTC.
And then again, on kernels = 2.6.22 you have hi-resolution timers which
generally work better.
I have yet to experience these, but it sounds promising.
Cheers
Tony
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branch, so won't appear in
a release until 1.6.1.
If you want to apply the patch yourself, you can find it in the bug tracker
at http://bugs.digium.com/view.php?id=11954
Cheers
Tony
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in doing
an x.y.z.a just because a change is small.
Cheers
Tony
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(${DIALSTATUS})
exten = h,1,NoOp(${QUEUESTATUS})
exten = h,2,NoOp(${DIALSTATUS})
Cheers
Tony
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to reach a priority after a gap is to Goto it in some way.
In your example above, if you return from priority 20, it will look only
for 21, and if not found, it won't keep looking for higher priorities.
Cheers
Tony
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to see the packets sent to/from
Asterisk.
Cheers
Tony
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, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Why is it doing so?
On Thu, Apr 17, 2008 at 2:36 AM, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED],
Rizwan Hisham [EMAIL PROTECTED] wrote:
I have been seeing a lot of the following warning messages on my
is with the carrier.
Cheers
Tony
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current call.
If he is the calling party, he will execute 'h' when he clears, but
if he is the called party, he won't be in the dialplan to execute 'h',
so we need some other way to invoke the ringback (step 5).
Thoughts?
Cheers
Tony
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out of retries.
Cheers
Tony
On Fri, 2008-04-04 at 10:35 +, Tony Mountifield wrote:
Has anyone here implemented Ring back when free in Asterisk?
The way it works in the UK is as follows:
1. A calls B. B is engaged (busy).
2. A hears The number you called is busy. To use ringback
wave files.
Thanks to all!
Jerry, glad you got it working. acpi=off did cross my mind, but I don't
really understand what it does, nor why it would help!
Cheers
Tony
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In article [EMAIL PROTECTED],
Michiel van Baak [EMAIL PROTECTED] wrote:
On 10:35, Fri 04 Apr 08, Tony Mountifield wrote:
Has anyone here implemented Ring back when free in Asterisk?
The way it works in the UK is as follows:
1. A calls B. B is engaged (busy).
2. A hears The number
like Call Waiting than Ringback When Free.
Cheers
Tony
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with Asterisk.
I do, frequently!
Cheers
Tony
Tony Mountifield wrote:
Has anyone here implemented Ring back when free in Asterisk?
The way it works in the UK is as follows:
1. A calls B. B is engaged (busy).
2. A hears The number you called is busy. To use ringback, press 5
3. A presses
to make the GUI compatible with IE
as well as FF?
Cheers
Tony
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in
asterisk-gui/branches/asterisknow and that trunk has lagged behind. Is that
correct?
Cheers
Tony
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In article [EMAIL PROTECTED],
Roderick A. Anderson [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
When I bring up the Asterisk GUI in AsteriskNOW, using IE7, it displays
a message at the top Your browser is not supported by this version of
GUI!,
and We recommend using Firefox
kernel version are you using? (uname -a)
Cheers
Tony
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In article [EMAIL PROTECTED],
Jerry Geis [EMAIL PROTECTED] wrote:
On Thu, Apr 03, 2008 at 01:45:09PM +, Tony Mountifield wrote:
/ Jerry, the first thing to check is cat /proc/interrupts and see if
there
// is an entry for rtc on IRQ 8. There should be, and the interrupt counts
- I've only ever used 32-bit.
Cheers
Tony
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asterisk
now?
Is this system used for other things, or could you try installing a
32-bit (i386) version of CentOS and trying again? 64-bit vs 32-bit
is the only think I can think of at the moment.
Also, what do you get from: fgrep RTC /var/log/messages
Cheers
Tony
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In article [EMAIL PROTECTED],
Tony Mountifield [EMAIL PROTECTED] wrote:
I have just installed ztdummy on a new system running 2.6.18-53.1.6.el5,
and it is incrementing fine. I didn't realise there was a newer kernel out;
I'll have to update and try again.
Just updated to 2.6.18-53.1.14.el5
, zaptel and asterisk should
automatically start.
Cheers
Tony
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hardware and buy yours.
It bothers me that a thread I started about a technical issue has been
used as a pretext for a commercial promotion that doesn't really address
the issues at hand.
Cheers
Tony
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the DIALSTATUS channel
variable. If the call was picked up, ${DIALSTATUS} will contain ANSWER.
Cheers
Tony
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On Wed, Apr 2, 2008 at 4:39 PM, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED],
Pete Kay [EMAIL
= 100,n,Read(OPTION,LONG-MESSAGE,2)
Cheers
Tony
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/bin/conf-background.agi. Try again with
0755 instead of 6644.
Cheers
Tony
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Tony
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To UNSUBSCRIBE
in place
of the hard coded number.
Cheers
Tony
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, and incrementing by 1
for each subsequent channel.
In Asterisk 1.4 or later, an optional system name can be defined in
asterisk.conf, and if defined, the unique ID becomes:
system_name-timestamp.seq_num
Cheers
Tony
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recording (r option), and MeetmeAdmin
operations such as mute all and unmute all.
Cheers
Tony
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for a system running
a small number of huge conferences. In the latter case, I want to look
at having a local conference on each box and bridging the conferences
together. This is the bit that gets complicated for making conf-wide
decisions :-)
Cheers
Tony
MATT---
On 3/4/08, Tony Mountifield
, but I'd be interested in any stories of success
or failure.
Cheers
Tony
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(${SAVEDEXTEN},1)
; allow user to press 6 to unmute all users
exten = 6,1,NoOp(unmuting conference ${CONF})
exten = 6,n,MeetMeAdmin(${CONF},n)
exten = 6,n,Goto(${SAVEDEXTEN},1)
Hope this helps!
Cheers
Tony
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of local is used.
Hope this helps!
Cheers
Tony
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asterisk
it is.
Cheers
Tony
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Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Feb 05, 2008 at 09:58:54AM +, Tony Mountifield wrote:
Specifically, what kernel version?
As I understand from an IRC chat - CentOS 5 - 2.6.18-something.
Hmm, ok.
If you are using 2.6.9 (e.g. Centos 4),
then by default when you build
In article [EMAIL PROTECTED],
Steve Edwards [EMAIL PROTECTED] wrote:
On Tue, 5 Feb 2008, Tony Mountifield wrote:
Tzafrir Cohen [EMAIL PROTECTED] wrote:
So that point is mute.
moot, not mute.
moot, not mute.
Now that point really *is* moot (i.e. open to debate, which is the true
)
exten = _X.,n,MeetMe(${EXTEN},dr)
Hope this helps!
Cheers
Tony
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.
Please give more details, and hopefully we can help you.
Cheers
Tony
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customisations or new
features I do create would still need to be ported to trunk before they
would have any chance of making it into Asterisk. This takes time, which
is always in short supply, and means that some cool features remain mine
only :-(
Cheers
Tony
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is still there in Trunk.
Cheers
Tony
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In article [EMAIL PROTECTED],
Kevin P. Fleming [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
When making or receiving a SIP call via my service provider, I get the
following message logged by Asterisk:
Dec 11 15:13:37 NOTICE[7392]: rtp.c:331 process_rfc3389: Comfort noise
support
on at the moment?
show channels
Cheers
Tony
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, by adding
half the divisor to the dividend before dividing. Without that, you just
get round down instead.
Cheers
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
In article [EMAIL PROTECTED],
Vincent [EMAIL PROTECTED] wrote:
On Mon, 12 Nov 2007 09:58:50 + (UTC), [EMAIL PROTECTED]
(Tony Mountifield) wrote:
I'm a little surprised at the variety of band-aid suggestions that have
been posted. All you need to do is refer to show application record
can press '#' to terminate the recording and continue to the next priority.
If the user should hangup during a recording, all data will be lost and the
application will teminate.
---
Cheers
Tony
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Tony Mountifield
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${ISDNCAUSE} on the next line of your dialplan.
Cheers
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
On Nov 12, 2007 2:04 PM, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
the conditions you are interested in. For example you might
want _X. to go on and try again or try a different operation that you couldn't
do within 'h'.
Many thanks for your informations!
Glad to help. Hope you get it working the way you want!
Cheers
Tony
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Tony Mountifield
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Thanks, Olle!
Cheers
Tony
In article [EMAIL PROTECTED],
Johansson Olle E [EMAIL PROTECTED] wrote:
7 nov 2007 kl. 14.26 skrev Tony Mountifield:
Quick question for those who know the innards of chan_sip:
Does chan_sip use the To: header of an incoming INVITE request,
for anything
In article [EMAIL PROTECTED],
Carles Pina i Estany [EMAIL PROTECTED] wrote:
On Nov/06/2007, Tony Mountifield wrote:
I have a number of systems that have a single Pentium 4 @ 2.8GHz (with HT),
1GB RAM and a 4xE1 PRI card (TE410P), and they regularly have conferences
with up to 90
. This in included in 1.2.19 onwards and 1.4.5 onwards.
See http://bugs.digium.com/view.php?id=9121 for details if needed.
Cheers
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
context
is taken from the section in sip.conf that matches the request's
source IP address. Is that correct?
Cheers
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
the required first part.
Any better ideas?
Thanks in advance!
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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with 15 users
even using VoIP and compressed codecs.
Cheers
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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In article [EMAIL PROTECTED],
Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Tzafrir Cohen [EMAIL PROTECTED] wrote:
As you can see, that script doesn't really redirect. It does not point
me to the new file name. If I use a web browser, I still get the
illusion
content is another matter altogether
Cheers
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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don't think you
can play a message to B, since B is not executing in the dialplan.
Cheers
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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needed . -f then? But this is the default of rm. The
shell got in your way?
Looks like one of the shell startup files might have rm aliased to rm -i
for all shells, rather than just for interactive shells. So the -f is
needed to override the -i.
Cheers
Tony
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Tony Mountifield
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in there about AEL, except a mention of extensions.ael on
page 471.
Cheers
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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