I am using version from git
5c840cf43838e0690873e73409491c392333b3b8 .
So, the question, how to fix, so I can get the tompile to work?
Thanks in advance for any suggestions.
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Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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have been using meetme and there you can just display the list of
users and you get that information.
Thanks in advance for any suggestions.
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cov...@ccs.covic
r joins first, it's met the criteria and will conference
> and start recording.
>
> Dan
>
> -Original Message-
> From: asterisk-users On Behalf Of
> John Covici
> Sent: Tuesday, June 28, 2022 6:28 PM
> To: asterisk-users@lists.digium.com
> Subject: [E
On Tue, 28 Jun 2022 19:54:11 -0400,
Joshua C. Colp wrote:
>
> [1 ]
> On Tue, Jun 28, 2022 at 8:28 PM John Covici wrote:
>
> > Hi. I have been using meetme for years, but I wanted to try
> > confbridge as meetme is going away soon.I am having a few
> >
ow do
you spend it?
John Covici wb2una
cov...@ccs.covici.com
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Check o
chan_sip, meetme or something else
entirely? Nothing relevant in the logs.
Thanks in advance for any suggestions.
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had multiple vlans configured for my servers (with voip exclusive to one).
>
> The short story is that it is easier to learn how to get things working
> within pjsip than learning the tricky networking setup.
>
>
> On Sun, Jan 9, 2022 at 2:49 AM Duncan Turnbull
> wrote:
>
> &g
On Sat, 08 Jan 2022 19:17:57 -0500,
Antony Stone wrote:
>
> On Sunday 09 January 2022 at 00:50:27, John Covici wrote:
>
> > Hi. I am using asterisk 18.3 and freepbx.
>
> Hm, which version of FreePBX uses Asterisk 18.3?
>
> > How can both sip and pjsip be list
ould like pjsit not to listen,till I figure out how to configure
the thing, so my logs don't fill up with messages.
Thanks in advance for any suggestions.
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>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
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>
his statements correct that we need some
service -- not necessarily his -- to sign the call before sending it
to our normal carrier, or will the normal carrier -- whoever -- sign
the call if they know the number?
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> Check out the new Asterisk community forum at: https://community.asterisk.org/
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> Doug
According to this document, there is no way for me to change the
volume(s) for another user, whereas meetme allows me to do this by
specifying the conference number and user number.
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ise I have to stay with meetme. And
I wonder if its a meetme issue?
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e of other
participants audio, meetme has this feature which I use frequently.
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New to Asterisk?
orum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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use Asterisk 16.
>
> On Mon, Oct 7, 2019 at 5:58 AM George Joseph wrote:
>
> On Fri, Oct 4, 2019 at 1:19 PM John Covici wrote:
>
> Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10
> system and I am running into the following problem. I need to install
&g
o
you spend it?
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an Floimair
> Innovation - Software-Development
>
> COMMEND INTERNATIONAL GMBH
> A-5020 Salzburg, Saalachstraße 51
> http://www.commend.com <http://www.commend.com/>
>
> Security and Communication by Commend
>
> FN 178618z | LG Salzburg
>
> Am 24.01.1
it does not work at all because it keeps
> telling in my log that modules support is not available, so no modules get
> loaded.
>
> Any ideas on thispuzzle would be appreciated.
>
>
> --
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ted.
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Hi. So, I applied the patch, works, but I could not figure out a
fail2ban regex which will hit that line, have you got one I can use?
Thanks.
On Thu, 30 Aug 2018 11:03:08 -0400,
sean darcy wrote:
>
> On 08/29/2018 09:33 PM, John Covici wrote:
> > OK, Thanks. I have a couple
ing list
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gt; On Thu, Aug 30, 2018 at 6:02 AM John Covici wrote:
>
> I agree, but is it possible to try over and over with anything other
> than the challenge warning in the security log as sean suggested and
> put a patch for?
>
> I don't think I understand your question.
>
/asterisk-users
> >>
> >> --
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risk-users mailing list
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> Check us out at: http://digium.com & htt
, 29 Aug 2018 20:31:15 -0400,
sean darcy wrote:
>
> On 08/29/2018 08:07 PM, John Covici wrote:
> > I wonder if I could have that patch, maybe I could add it to my
> > fail2ban regexp and if you have the correct regexp, I would apperciate
> > that as well.
> >
>
>
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Also, make sure you are using fail2ban and that you have good
passwords on your extensions.
On Fri, 28 Oct 2016 11:55:42 -0400,
John Covici wrote:
>
> How about a \ before the - ?
>
> On Fri, 28 Oct 2016 11:38:13 -0400,
> Markus wrote:
> >
> > Hi list,
> >
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eplaced by ConfBridge on all currently supported
> Asterisk
> versions.
>
Except that confbridge lacks some features that meetme has -- I wish
this were not so.
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those
files?
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ed.
> >
> > --
> > Your life is like a penny. You're going to lose it. The question is:
> > How do
> > you spend it?
> >
> > John Covici
> > cov...@ccs.covici.com
> >
> > --
> >
rver to do the job for you.
>
> --
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How do
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t;
> > sean
> >
> >
> >
>
> Thanks for the reply, but any 'w' s in the dial string cause
> CHAN_UNAVAILABLE.
>
> I'm not sure I'm up for learning agi just yet. I was hoping for a
> dialplan solution.
>
> sean
Those W&#x
nd it?
John Covici
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ather than
reject the call? Is there a way to do this correctly as far as asterisk
is concerned?
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in if
none of them match one that asterisk likes.
If I only offer one codec, it works, but that seems an unnecessary
restriction to me.
Any assistance on this would be appreciated.
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the timing from an installed card if available,
> but now it is smart enough to detect if there is not a card
> installed or operating properly and still provide timing without
> requiring the user to load "dahdi_dummy" explicitly.
You are correct, when meetme didn't work,
built-in timer, so there is no need for a
> separate module. The ChangeLog entry above is correct, as of DAHDI 2.4
> and later.
So, how do I get this to work -- when I tried to do this, I could get a
conference all right, but it would not record the con
-
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I would appreciate it if someone could take a look and tell me what can
be done to fix this problem.
Thanks in advance for any ideas.
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> >>
> >> Thanks in advance for any ideas.
> >
> < You installed libpri ?
> >>I don't have any pri's.
>
> I'd check the dmesg output - AFAIK you need libpri as a backbone for DAHDI
> (at least on some kernels).
No dmesg out
d it looks like it did -- the dahdi-version.h has a time stamp about 2
minutes before the timestamp of the modules in the kernel I was using.
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ools-trunk 8670.
>
> How can I get this to work correctly?
>
> Thanks in advance for any ideas.
>
> You installed libpri ?
I don't have any pri's.
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64 Feb 17 11:12 8881-
> > conf-20110217-100052.wav
> > -rw-r--r-- 1 asterisk asterisk 66971884 Feb 17 11:11 8881-
> > conf-20110217-100117.wav
> > -rw-r--r-- 1 asterisk asterisk 66648684 Feb 17
g works as expected.
Not so, I always run ./configure as root and I get the message that
32768 exceeds ...
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'module reload chan_dahdi.so' is not the same as:
>
> module unload chan_dahdi.so
> module load chan_dahdi.so
>
> Please try that one.
>
> >
> > The output of dahdi show channels is just the header line with no
> > channels.
>
> OK. I
#x27; ?
I did module reload chan_dahdi.so but no joy.
The output of dahdi show channels is just the header line with no
channels.
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gt;
> >
> > Now chan_dahdi is indeed loaded, but I have an empty chan_dahdi.conf.
>
> Not even an empty '[channels]' section?
I did put that just now, but I still get the same warning.
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debug this one?
> >
> > Any ideas would be appreciated.
>
> What version of DAHDI?
PProblem solved by Shaun fixing a regression in dahdi-trunk. Thanks to
him its now working.
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in this conference,.
How would I even start to debug this one?
Any ideas would be appreciated.
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t;>> description:DAHDI Telephony Interface
> >>> author: Mark Spencer
> >>> srcversion: A63E42F5ADDDE39777BCC24
> >>> depends:
> >>> vermagic: 2.6.26-2-686 SMP mod_unload modversions 686
> >>> parm:
table [crc_ccitt]
> >>>
> >>>> ]# modinfo dahdi
> >>> filename: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko
> >>> version:SVN-trunk-r9614
> >>> alias: dahdi_dummy
> >>> license:GPL v2
> &g
rc_ccitt_table [crc_ccitt]
> >>>
> >>>> ]# modinfo dahdi
> >>> filename: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko
> >>> version:SVN-trunk-r9614
> >>> alias: dahdi_dummy
> >>> license:GPL v2
6.26-2-686/dahdi/wctdm.ko:
/lib/modules/2.6.26-2-686/dahdi/dahdi.ko
/lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko
/lib/modules/2.6.26-2-686/dahdi/dahdi_echocan_jpah.ko:
/lib/modules/2.6.26-2-686/dahdi/dahdi.ko
/lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.
SVN-trunk-r9614
alias: dahdi_dummy
license: GPL v2
description:DAHDI Telephony Interface
author: Mark Spencer
srcversion: A63E42F5ADDDE39777BCC24
depends:
vermagic: 2.6.26-2-686 SMP mod_unload modversions 686
parm: debug:Sets debugging verbosity as
system with the new modules, but no joy at all. Igot the same results
whether I rebuilt the kernel or not, so this is what is confusing to me.
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.
Any ideas would be appreciated.
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isk.org/hello
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r and see if they hear
the tones.
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up to a point and then they "fixed" something.
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self, but I still want
to hear the other end and asterisk is cutting off both ends audio.
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oes silence suppression.
>
But it surpresses in both directions! I still want to hear the other
end. For a test is there a way to turn off that feature to see if that
is the cause?
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is that what you need? I did do
an rtp set debug and this is how I found out that when I hit the mute
button on the soft phone all rtp traffic ceased between the phone and
the asterisk box.
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this does not happen in that vrsion, but right after that this
strange thing starts and is not fixed in the current one.
Any assistance here would be appreciated.
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way the last good install for me is 281875 so it
right after that where from express talk to an outside line through
asterisk is failing with one way audio after the first several seconds.
I did try latest update and it is
ne next in the logs, maybe I will try latest and see if it
goes away.
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n the commit?
Its the 282911 commit seems to break audio to the soft phone, but not
to my ata -- very strange.
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phone, but not the other way around. This looks
like one commit, but obviously I would like to know what's going on
here?
Thanks in advance for any ideas.
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ence menu plays and all is well except -
>
> 2. According to the instructions, pressing 8 should leave the conference.
> No dice.
8 exits the menu, not the conference.
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for the *1
> >
> > [Jul 29 20:31:03] VERBOSE[22300] file.c: --
> > Playing 'beep.gsm' (language 'en')
> > [Jul 29 20:31:04] VERBOSE[22300] file.c: --
> > Playing 'beep.gsm' (language 'en')
> > Why the s
was answered and after the
dtmf for the *1
[Jul 29 20:31:03] VERBOSE[22300] file.c: --
Playing 'beep.gsm' (language 'en')
[Jul 29 20:31:04] VERBOSE[22300] file.c: --
Playing 'beep.gsm' (language 'en')
Why the second line?
problems.
Should I file a bug, or is this going to be fixed?
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-- Bandwidt
clarification.
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x14036
^
After the "" was gone in the set, it works in 1.8, have not tried in
1.6.
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an unoptimized backtrace.
That fixed my module reload seg fault as well.
Thanks.
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Your life is like a penny. You're going to lose it. The question is:
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John Covici
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alue.
> >
> Post an example for working (1.6) and not working (1.8)
>
OK, the line actually is:
exten => s,n(auth),Set(password="") which sets to a null value in 1.6.2
but does not in 1.8.
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