said
they were running CentOS, and both were doing kludges with individual
modprobes (in one case TWICE!) instead of just using the supplied
script.
Cheers
Tony
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, but the TE412P is £200 more expensive.
Is there any reason it would be worth paying the extra for the 412?
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called README.udev
Cheers
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want
to direct the PRI debugging to a file with pri set debug file filename.
It's not clear from the log you posted whether q931_hangup() was called
because of a Q.931 message Asterisk received, or just because it decided to.
Hopefully, the intense debug would make that clear.
Cheers
Tony
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length of time.
Just a couple of ideas to try.
If the phone is still sending RTP when muted, the stream presumably contains
perfect silence. I don't know if that gets detected anywhere.
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Tony
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.
To do so, you need to duplicate the entries with _X
It's a pity that . means one-or-more-chars rather than zero-or-more-chars.
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]
-=-=-=-=-=-
-=-=-=-=-=-
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Interesting. Digium's the only one in the list that I've heard of!
BTW, don't read anything into being second on the list - the companies
are listed in alphabetical order.
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Tony
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the asterisk logs give any clues about why they are dropping the calls?
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is nothing instead of 1.
How can I maintain the value of the variable in the above case?
Try using Set(__testmode=1), but still refer to it as ${testmode}.
The __ tells Asterisk to propagate the variable to created channels.
Cheers
Tony
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In article [EMAIL PROTECTED],
unplug [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
Try using Set(__testmode=1), but still refer to it as ${testmode}.
The __ tells Asterisk to propagate the variable to created channels.
Do you mean it is a global variable instead of channel variable
this:
Action: Originate
ActionID: whatever
Channel: Local/[EMAIL PROTECTED]
Context: outgoing
Exten: 1357924680
Priority: 1
Async: true
Then watch out for the OriginateSuccess or OriginateFailure event.
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Tony
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Play
/pipermail/asterisk-users/2006-August/162582.html
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on that.
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Hope this helps!
Cheers
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format you like using sox, e.g. for .wav:
samples.pl 50 busy.h | sox -t raw -sw -c1 -r8000 - busy.wav
samples.pl 100 ringtone.h | sox -t raw -sw -c1 -r8000 - ringtone.wav
Change .wav to .ul or .al for uLaw or aLaw output.
Hope this helps!
Cheers
Tony
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optimise themselves out when the call is set up.
Hope this helps!
Cheers
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back to the manager application just before doing
the Dial.
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that you generate...
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this really reliable is to use a digital interface,
either BRI or PRI.
Hope this helps
Cheers
Tony
On 8/8/06, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
shawn bright [EMAIL PROTECTED] wrote:
i have a python script that drops a call.call file into the
/var
Silence Suppression in
some cases.
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Silence Suppression or VAD
on the Cisco for the link to your Asterisk box.
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Tony
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.
Is this a reasonable amount of calls to handle with meetme?¿?
That depends on many factors: CPU power; whether the calls come in via
a quad-E1 card or VoIP; for VoIP, what codec is used; etc.
Cheers
Tony
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channels
conferencing among them
only on zap devices with a PIV/ 3GHz 1GB Ram?
I think you should manage it easily, based on my experience with similar setups.
Cheers
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8khzfile.wav -a adpcmfile.wav
This gives a factor of 4 reduction, from 16 bits per sample to 4 bits
per sample, i.e. 4000 bytes per second, plus 44 byte header.
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Tony
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as:
SIP/dundisip:[EMAIL PROTECTED]@xxx.yyy.142.163
See the following line in the sample extensions.conf as an example:
;exten = _42X.,1,Dial(SIP/user:[EMAIL PROTECTED]:[EMAIL PROTECTED],30,rT)
Cheers
Tony
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In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 2:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: DUNDi with SIP
In article
an AGISTATUS
variable instead of just a return valoue of -1.
What is the best way to approach this?
Cheers
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In article [EMAIL PROTECTED],
Rich Adamson [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
Using the 1.2 branch of SVN, I've been experimenting with FastAGI.
I want to do something useful for the caller (e.g. play a message)
if the FastAGI server is not running, i.e. AGI gets connect
there were originally two calls to manager_event(),
one for OriginateFailure and another for OriginateSuccess.
They have now been combined into one, with a conditional event name,
which may have given rise to the mistaken impression if just skimming
the diffs cursorily.
Cheers
Tony
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In article [EMAIL PROTECTED],
Tony Mountifield [EMAIL PROTECTED] wrote:
[... lots of stuff about trying to register two numbers in the same
SIP domain with different proxies ...]
Are all the SIP experts on vacation this week then?
I'd appreciate any help with my original posting if anyone here
of rw_lock_t in other kernel
versions, this doesn't really need to be made version-dependent, does it?
Could just do this and it would be compatible with any version:
CFLAGS+= -Drw_lock_t=rwlock_t
Cheers
Tony
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, but only if it will help
with this issue.
Thanks in advance,
Tony
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you need to do GET VARIABLE DUNDILOOKUP(2944093|180net), with
no ${}, and then do whatever you need with the return value, such as
passing it to SET VARIABLE DIALPATH
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Tony
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uses stdout to talk to
Asterisk.
However, to reply to Doug, if you want to log the stderr with timestamps
via syslog you could do it by piping through logger (man logger):
open STDERR, | /usr/bin/logger -p local0.notice -t AGI;
Cheers
Tony
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version of Zaptel are you using, what version of Asterisk, and
which Linux kernel does your system have?
Cheers
Tony
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or uses the name file, but retain the code that plays
the has joined or has left files. Change those file names to something
of your choosing, and record the messages you desire into those files.
Hope this helps
Cheers
Tony
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we return
DIALSTATUS instead, and do a GOTO based on that.
I've probably missed a bit, but certainly I don't think there was the
idea that everything should become the same extension and just use labels.
Cheers
Tony
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Play
to gather debugging information.
Cheers
Tony
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In article [EMAIL PROTECTED],
Andrew Furey [EMAIL PROTECTED] wrote:
On 6/1/06, Tony Mountifield [EMAIL PROTECTED] wrote:
# script /tmp/output.txt
Script started, file is /tmp/output.txt
# exec asterisk -rv
... do asterisky stuff ...
host*CLI exit
Script done, file is /tmp
as a non-root user, then it
is necessary to chown the call file to the user under which asterisk
runs, before moving it into the outgoing directory. Or to chmod it
to 666.
Cheers
Tony
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will also need to change signalling=pri_cpe to signalling=pri_net
in zapata.conf and restart asterisk.
Cheers
Tony
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rotate.
Cheers
Tony
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In article [EMAIL PROTECTED],
Steven Totaro [EMAIL PROTECTED] wrote:
Exten = h,1,hangup ?
No, there's never any need to call Hangup in the h extension, because
by the time h is called, the call is already hung up, by definition.
Cheers
Tony
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. Then the strcmp will never match:
} else if (!strcmp(exten, x*0)) {
Cheers
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this, but it is definitely of the
add-on variety.
You can do it pure dial plan if you are willing to have a menu that says
Press 1 to join as admin, Press 2 to join as participant.
Or you just define two different extensions - one for the admin to dial
and one for normal users.
Cheers
Tony
--
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file
formats:
http://www.astlinux.org/index.php?option=com_contenttask=viewid=38Itemid=43
In my opinion, any new Asterisk sounds should be put in the tree in four
formats at least: .wav, .al, .ul, .gsm
Cheers
Tony
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Play
-asynchronous.patch
Then apply it by hand to channel.c, and also to app_milliwatt.c and app_sms.c
if you happen to be using those applications. I don't think the app_chanspy.c
patch is required any more.
Cheers
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even if not documented.
Are you certain the version in which you had the 'R' option was standard
Asterisk? Could it have been a BRIstuff version instead? Or with some other
patch?
Cheers
Tony
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I could probably do it with a separate supervisor program using the
Manager API, but can't think of any way to do it with AGI.
Cheers
Tony
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.
The other situation to take account of is when the caller somehow adds
to his prepaid balance while he has one or more calls in progress, in
order to avoid being cut off during the call.
Cheers
Tony
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In article [EMAIL PROTECTED],
Jon Farmer [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
The other situation to take account of is when the caller somehow adds
to his prepaid balance while he has one or more calls in progress, in
order to avoid being cut off during the call.
Yes
automatically.
Cheers
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, but I like the idea of it being done per-call on demand
using AGI if possible.
Can anyone suggest any ideas or better techniques?
Thanks in advance!
Tony
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the patch from http://bugs.digium.com/view.php?id=3405
It's quite old, so you will most likely have to apply it by hand.
Cheers
Tony
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bug #5697.
That dynamic buffer patch is not really required. It was an attempt to
get the pseudo-device to accept a whole frame in one write(), but this
has been obviated by the non-blocking flag to careful_write().
Try updating to an unpatched 1.2.7 instead.
Cheers
Tony
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,Set(INITIALCONTEXT=${CONTEXT})
exten = ... and so on
[somewhere-else]
Dial(Local/[EMAIL PROTECTED])
Or something like that.
And couldn't you use Goto(${INITIALCONTEXT},${NUMBER},1) instead of
Dial(Local/[EMAIL PROTECTED]) ?
Cheers
Tony
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, 8000Hz is more useful,
since it contains the same audio data as .raw or .sln, but with the
addition of a 44-byte header that allows you easily to play it with
standard utilities on either Windows or Linux.
Cheers
Tony
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Play
it contains the same audio data as .raw or .sln, but with the addition of a
44-byte header that allows you easily to play it with standard utilities on
either Windows or Linux.
Cheers
Tony
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from GSM.
Cheers
Tony
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the line and wait a litle bit before
calling MeetMe:
exten = 6000,1,Answer
exten = 6000,2,Wait(0.5)
exten = 6000,3,MeetMe(1234|ciMp)
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, so I would be happy with a permanent
change of behaviour.
Cheers
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interface better than the latest AAH.
Presumably, the next version of AAH will include FreePBX.
Cheers
Tony
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. This is on a recent version of Asterisk. Any comments/suggestion?
Make sure you are specifying it as /n, and not \n like you wrote above.
If it still doesn't work, then it is most likely a bug...
Cheers
Tony
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similar techniques.
Cheers
Tony
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minute.
It then becomes a question of whether it is cost-effective to strive for
even better than that.
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would be in /var/log/maillog on the box.
Look particularly around the time when a voicemail should have been sent.
Cheers
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(top posting for brevity, but original post included below, as it was
over seven weeks ago)
I've at last updated the patches for both trunk and 1.2, and posted them
to Mantis at http://bugs.digium.com/view.php?id=6731
Cheers
Tony
In article [EMAIL PROTECTED],
Tony Mountifield [EMAIL PROTECTED
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for transferring both legs of the call into a MeetMe conference.
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be replicating the same
functionality, and making sure it does not feed back audio from the
speakers.
I think It would be much more difficult to try to do it at the asterisk
end.
Cheers
Tony
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of setup?
Cheers
Tony
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in the 'h' extension.
If it's broken, then that's a bug.
However, if your example is an exact copy from your dialplan, perhaps
the parser is getting confused, because {$service} should be ${service}
Cheers
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/listinfo/asterisk-users
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In article [EMAIL PROTECTED],
scott [EMAIL PROTECTED] wrote:
Hi
Does anyone know a UK Voip Proivder that will give me more than 1 telephone
number and point
it to my sip account.
www.voiptalk.org
They do both SIP and IAX and you can buy as many numbers as you need.
Cheers
Tony
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.
---- --- - - - -- - - -- - - - --- - -- -
- --- - - -- -
-- -- - --
Tony Mountifield [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
In article [EMAIL PROTECTED],
Steven [EMAIL PROTECTED] wrote:
This is great news.
Agreed!
Previously, stable was just considered a snapshot and if you ran
stable
In article [EMAIL PROTECTED],
Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Dan Austin [EMAIL PROTECTED] wrote:
Tony wrote:
I should tidy it up and submit it, but haven't got round to it :-(
Let us know if you can. I'm already maintaining a grocery list
) in new
stack
with this above line ^^^
There is a parameter missing between fetchid and vm_active. Note the two
spaces. I suspect you might have a typo in the variable substitution for
${resultid}.
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applied to the
stable 1.0 branch. Where you needed to use head was if you wanted to
use the cool new features that were never going into 1.0.
Cheers
Tony
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that message, or
- replace the sound file conf-onlyperson.gsm with a file of zero duration
(not necessarily just an empty file - it might still need a header).
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Tony
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Tony Mountifield
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think you can. The G option was to put you both into the MeetMe
room at the beginning of the call.
I think the solution needs a little more thinking about
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Tony
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, but one of them would contain the dialled extension
instead of the last extension processed for the call.
Cheers
Tony
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.
I should be able do so this weekend. That's the plan, anyway :-)
I'll post the Mantis bug# when I've submitted it.
Cheers
Tony
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that an 'l' flag meant
listen-only (like the current 'm'), and an 'm' flag meant initially-muted.
I also put in Manager Events to inform when a user was muted or unmuted.
I should tidy it up and submit it, but haven't got round to it :-(
Cheers
Tony
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Tony Mountifield
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the figures, and use ztdummy with USE_RTC without
the patch from 5971. Then just describe any operational effect you might
be seeing, if any.
Thanks in advance,
Tamas
Cheers
Tony
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Tony Mountifield
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it to get
the old wrong value out of their cache.
Cheers
Tony
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Asterisk
at it, remove the ones you don't need, too.
Cheers
Tony
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including, even though it is not in the
standard 1.2.1. That does give scope for confusion though!
However, it is a very worthwhile patch, and I hope it does make it into
1.2.2, not just the trunk, which will be months away from being
recommended for production.
Cheers
Tony
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Tony Mountifield
a check in there and a printk for unexpected
sequence numbers.
Cheers
Tony
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to terminate the
sequence.
Hope this helps
Cheers
Tony
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, it jumps to notfound because NUMD is not set.
Your dialplan can have as many possible steps as you want, it's not expensive.
Cheers
Tony
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with Cookies must be enabled!.
But I never disable cookies, and other cookie-using sites work fine.
Cheers
Tony
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the database transactions necessary to decide
on the disposition of the call, and then jumps to the appropriate
extension and priority in the dialplan.
Cheers
Tony
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Tony Mountifield
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is complaining about the empty message (NUL byte) that is
immediately after the final \r\n\r\n.
You also don't need to use two different buffers: you can re-use the same
one for the second message.
Cheers
Tony
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probably omitted to perform the steps it describes.
Cheers
Tony
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Tony Mountifield
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a way. I did look at the code, and I don't think it's
possible without changes (e.g. to convert | or \n to newlines).
BTW, please post to the list in plain text, not HTML.
Cheers
Tony
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and then specific allow lines for the codecs you want
to support, such as alaw, gsm and so on.
Don't forget a reload in order to act upon the changes.
Cheers
Tony
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Tony Mountifield
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.
If not, does anyone have any pointers on what would be required to
add a new signalling type with the behaviour described?
Thanks in advance
Tony
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