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this. The agents.conf is all commented out except for [general] .
Anyone know what is happening?
Thanks.
P.S. I deleted the modules directory before installing.
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upgrading to
svn-275706 from an earlier version 262801. Its a certain argument of
meetme which I have not trafcked down yet which is causing this.
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, but if it is conference number, cMs then it dies -- asterisk
dies. Is this enough for someone to figure out?
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run in
verbosity 4, but am not seeing anything of interest.
Any ideas would be appreciated.
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to troubleshoot. I usually run in
verbosity 4, but am not seeing anything of interest.
Any ideas would be appreciated.
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ideas would be appreciated.
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You need in your ./configure command line --prefix=/usr
--mandir=/usr/share/man along with any other options and asterisk should
see things.
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netjet and now things are working.
But I wonder what is going on here and where did netjet come from -- it
doesn't look like an dahdi module to me.
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the old version would not compile
under 2.6.32, and update of dahdi-tools-trunk make no difference.
Any ideas would be appreciated.
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certainly did not do it myself -- I
thought modprobing the modules would do such a thing. I will certainly
try lsdahdi when I can reboot into the new kernel.
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can tell.
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the command line
completely outside of Asterisk.
OK, are there include files available for the appropriate functionality?
Sounds like it might be very nice.
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the phone, but
you'd almost have to have an audio display of user numbers and caller
ids to have it make sense.
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and look at the log and
lsmod after you do that.
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John,
modprobe comes back right away with ztdummy not found
[r
You should not have removed all kernel modules, at least zaptel, but I
keep them anyway.
on Monday 05/18/2009 Jimmy Ezell(jez...@hmhca.com) wrote
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of
John covici
Sent: Monday, May 18
covici
Sent: Monday, May 18, 2009 05:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
This is your problem -- check the compile of the zaptel package and
see what happened. It might not compile
, 2009 at 05:32:45PM -0400, John covici wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
short and some IVR's are unhappy or don't detect it. I did poke
around, but it looks like when RFC2833
OK, thanks. If I could convince them to use info, would that be
better as far as the duration is concerned?
on Monday 04/13/2009 Brent Davidson(br...@texascountrytitle.com) wrote
John covici wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like
on Tuesday 04/14/2009 Kristian Kielhofner(kristian.kielhof...@gmail.com) wrote
On Mon, Apr 13, 2009 at 5:32 PM, John covici cov...@ccs.covici.com wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
translation or
where it pulls the toneduration setting from if no PSTN interface is
involved in the call.
-Brent
John covici wrote:
OK, thanks. If I could convince them to use info, would that be
better as far as the duration is concerned?
on Monday 04/13/2009 Brent Davidson
/view/Asterisk+cmd+SIPdtmfmode
John covici wrote:
Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4.
Is this new in 1.6?
on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote
To the best of my knowledge, the only way for you to control
I have no idea how to increase that
duration.
Any assistance would be appreciated.
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If I revert back to my old version of asterisk, it works just fine.
on Thursday 04/09/2009 Matt Riddell(li...@venturevoip.com) wrote
On 4/04/2009 2:22 a.m., John covici wrote:
The minute asterisk tries to execute an agi, it gets utils.c write
error broken pipe and so hangs up the call
to lose it. The question is:
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phones you do Transfer button/dial number/hear
parking slot/Transfer button again/Hang up.
John covici wrote:
I do the following from the legacy phone: hit theflash and get a
dialtone from the call, dial 70, the call is parked, but hangs up from
me immediately -- isn't this an attended
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OK, thanks.
on Tuesday 11/04/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
On Tue, Nov 04, 2008 at 03:17:30PM -0500, John covici wrote:
Hi. I am using gentoo kernel 2.6.27-r2 and dahdi trunk svn 5211 and
it will not compile with this kernel whereas it does compile with
2.6.25. Here
it?
John Covici
[EMAIL PROTECTED]
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=from-pstn
signalling=fxs_ks
faxdetect=incoming
rxgain=4.0
callwaiting=yes
channel=4
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Any assistance would be appreciated.
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Sent: Thursday, September 11, 2008 11:47 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] dahdi vs zap with latest version of asterisk --
having some problems
Hi. I am using asterisk 1.4 branch svn from yesterday and I am having
some problems -- I am still using
anonymous' has no member named
'ifru_settings'
I previously installed dahdi-trunk before trying to compile this
package, but no joy.
Any assistance would be appreciated.
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on Sunday 09/07/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
On Sun, Sep 07, 2008 at 09:22:57AM -0400, John covici wrote:
Hi. I am getting the following error while trying to compile the
dahdi-tools-trunk from svn this morning.
gcc -g -O2 -I. -O2 -g -fPIC -Wall -DBUILDING_TONEZONE
it?
John Covici
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OK, thanks.
on Friday 09/05/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
On Fri, Sep 05, 2008 at 10:32:30AM -0400, John covici wrote:
Hi. I want to use the new asterisk 1.4 with dahdi, but I would like
to know the svn branches for the dahdi, so I can use them that way --
much easier
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Hi. I have a number of sections in my iax configurationwhich was
generated by freepbx like this:
[covici]
username=covici
type=friend
secret=password
host=dynamic
disallow=all
context=from-internal
allow=ulaw
[Carol 0326]
host=iax.binfone.com
username=3234740326
secret=another password
type
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life is like a penny. You're going to lose it. The question is:
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change on my side to get this working ?
Jared,
I thought MixMonitor() was for Queue, Can you let me know how to use it?
Thanking you for replying.
Regards,
Sanjay Rajdev
- Original Message -
From: John covici [EMAIL PROTECTED]
To: [EMAIL PROTECTED
. You're going to lose it. The question is:
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Yep, you guessed it, an activvoice system. Anyway to make Asterisk
act like that for a while?
Thanks.
on Friday 04/11/2008 Doug([EMAIL PROTECTED]) wrote
At 17:32 4/11/2008, John covici wrote:
Hi. One of my clients has an old Mitel SX 200 with a separate
computer doing the voicemail
and receive information needed for a voice mail
system. With FXO/FXS ports is not possible receive such information.
Jorge Mendoza
John covici wrote:
Yep, you guessed it, an activvoice system. Anyway to make Asterisk
act like that for a while?
Thanks
If I see this, then messages are getting through.
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On Tue, Feb 26, 2008 at 06:45:15AM -0500, John covici wrote:
Hi. Since I am stuck with kernel 2.6.24, is there any way to compile
zaptel 1.4.7.1 under kernel 2.6.24? I tried using
make KBUILD_NOPEDANTIC=1 -- however this does not compile. Any other
suggestions for this and can I
I do have the newer one installed -- is this the problem?
And why this weird error anyway?
on Tuesday 02/26/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
On Tue, Feb 26, 2008 at 07:50:09AM -0500, John covici wrote:
I am getting this strange error:
make[1]: Entering directory `/usr/src
being changed in the Makefile. So how
can I compile this thing till there is a fix for the bug or the
regression is removed?
on Tuesday 02/26/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
Slightly edited your message:
On Tue, Feb 26, 2008 at 08:48:13AM -0500, John covici wrote:
on Tuesday
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I updated zaptel and I can dial out, but when someone calls in it
won't hangup unless my extension hangs up, which was not true before.
This is a better state than before, thanks much for fixing so far.
on Tuesday 02/26/2008 Shaun Ruffell([EMAIL PROTECTED]) wrote
John Covici wrote:
Hi. I am
it. The question is:
How do
you spend it?
John Covici
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How would I use that with kernel 2.6.24?
on Monday 02/25/2008 Shaun Ruffell([EMAIL PROTECTED]) wrote
Hi John,
John Covici wrote:
Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4
(latest as of today) and I cannot dial out using my 400P card with one
fxs module
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John Covici
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-users
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this, do any of the channel banks have this capability?
Thanks.
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John Covici
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OK, to clarify a bit, he wants to fix things so that all we are
depending on are the pots lines -- I know if they go out you are
gone. So what can we do in that case?
on Wednesday 01/02/2008 Tilghman Lesher([EMAIL PROTECTED]) wrote
On Wednesday 02 January 2008 17:10:05 John covici wrote:
Hi
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