are not on site). I would much
prefer for asterisk to keep running with what it has got.
I will be looking into the code (and this might be fixed in cvs-head),
but I would like to start a discussion on this first.
Thanks,
Jeb
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Jeb Campbell
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to start and have
parts of the dialplan fail, rather than have asterisk not load at all.
As I said, I have not checked the behavior of cvs-head, I just wanted to
discuss making asterisk more resilient.
Thanks for the tip and I will look into it.
Jeb
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Jeb Campbell
[EMAIL PROTECTED
be able
to limp along. What would you think of a commercial phone system that
completely dies when one port dies?
I appreciate your points (thats why I wanted to discuss this), maybe we
could satisfy both with an option in zapata.conf (keepgoing=yes)?
Thanks,
Jeb
--
Jeb Campbell
[EMAIL
,Hangup
Really easy to modify. Have fun.
Again this could be cleaner, but I got that other script working and
haven't had the time or need to clean it up.
Jeb Campbell
[EMAIL PROTECTED]
#!/usr/bin/perl
#
#
# allcall.agi will add all your Polycom sip phones to a meet me
# conference for use in office
info.
Jeb Campbell
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does not answer,
the redirected phone's voicemail plays and not the companies.
I just wanted to see if anyone else had this problem (and a solution).
Jeb Campbell
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http
the locations, you
get a 4 port TDM card (2 FXO, 2 FXS) at each location and hook into the
avaya. You could even setup asterisk and the avaya to pass the digits
back and forth for extensions (but unless you do it yourself, it will
cost a bit to setup).
Good luck,
Jeb Campbell
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interested in this, and even a timeframe for
availability would be great news.
Jeb Campbell
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verify this for yourself **
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it starts the count). When I
was having problems, I was getting hundreds of irq misses a minute.
For zttool to compile you must have newt installed (or libnewt
libnewt-devel for rpm distros -- something like that anyway). Hope that
helps.
Jeb Campbell
[EMAIL PROTECTED
, but then went to glibc -- and now have a full
system (perl, python, etc) in ~57M.
Anyway I was just wandering if you had your build sources/scripts online
so that people could customize Astlinux? I for one would like to be
using the stable cvs.
Thanks,
Jeb Campbell
[EMAIL PROTECTED
) solution to this.
Jeb
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-12
Thanks for any tips,
Jeb
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to think of it, I could use your patch to effectively disable the
interval, and use a cron script to restart asterisk nightly.
Thanks again,
Jeb
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Matthew Boehm wrote:
Thats not even the newest firmware, 7.2 is newest. Isn't this illegal?
Matthew
Definitely illegal, but 7.3 is the latest SIP firmware.
Jeb Campbell
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,
and 2 gigabit.
Jeb Campbell
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(assuming it was analog and
autoanswered) but no luck.
I would be happy to replace if anyone knows of an analog phone to page
system, but of course I would like to reuse what is there.
Thanks for any advice or pointers,
Jeb Campbell
[EMAIL PROTECTED
which
polarity they want). No adapters needed for either. And it is about $1100.
We are using 4 of them and love them.
Jeb Campbell
[EMAIL PROTECTED]
Christopher L. Wade wrote:
I know this is on the wiki, I just want to confirm so I don't blow up my
cisco phones. I've got several cisco 7940's
Just wanted to say Thanks to the Asterisk community -- all links are
bookmarked now!
Jeb Campbell
[EMAIL PROTECTED]
On Sep 21, 2004, at 4:54 PM, John Hill wrote:
Her is the 7905-12 page
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905
-Original Message-
From: [EMAIL PROTECTED
Hello all,
I feel dumb asking this, but does anyone have a link to the SIP
firmware for the 7912 on Cisco's site?
I have a SmartNet contract, but I just can't find the link (you can
search for 7960 sip firmware and find that fast).
Thanks for the help,
Jeb Campbell
[EMAIL PROTECTED
On May 2, 2004, at 1:02 PM, Ryan Courtnage wrote:
I'm personally using it with spandsp and having no problems, but YMMV.
If you want to enable it, goto line 60 of dsp.c and uncomment that
#define so it looks like this:
/* Define if you want the fax detector -- NOT RECOMMENDED IN -STABLE
*/
isp is 201!
In short, if you want more hosts get your isp to give you more
addresses.
Jeb Campbell
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is getting DTMF because my menu in extensions.conf works.
I will go through the code Friday, but I just didn't know if anyone
else was seeing this?
Setup: 23 voice pri from Avaya PBX to Asterisk IVR.
Thanks,
Jeb Campbell
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On Apr 7, 2004, at 4:23 PM, Darren Nay wrote:
My question is. Is there a way to make asterisk aware of the
calling-from (callerID) number so that it will automatically detect
the number and then go directly to asking them to input their
password.
From show application VoicemailMain try:
(as
long as your phone person is fast at setting up the routes).
Anyway, if you have questions, feel free to email me -- I want to pass
on the knowledge that so many here have helped with.
Jeb
Jeb Campbell
[EMAIL PROTECTED]
Cell: 865-385-1437
is internal and I understand not opening up systems,
but maybe digium would host this if you cannot as this is a HUGE
feature for asterisk.
Thanks,
Jeb Campbell
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Cell: 865-385-1437
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a current config).
So to all about to do this, I wish you better luck than I have had.
I'm sure if you have more Avaya experience you will be fine; I just
can't get in there and try things. I will get this working and post
configs.
Jeb Campbell
[EMAIL PROTECTED]
Cell: 865-385-1437
On Mar 29, 2004, at 11:06 AM, Eric Wieling wrote:
Jeb Campbell wrote:
Anyway, the only stuff off list was trying to debug the connection.
1. With a crossover there is no sync (YELLOW and RED alarms)
2. With standard cable I get a pri error that they think they are the
NET, but we are the NET
also, and flawlessly. I think you are right about
that involving libtiff.
I'm running 3.5.7-r1, but I changed the ebuild. If you change the
ebuild from make ... install to
make ... install-private, you get the headers you need (it's in the
function src_install).
Anyway that's my 2 cents,
Jeb
('',$ARGV[0]);
Now they are split!
Jeb Campbell
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not tried these myself.
Jeb Campbell
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= s,106,System(/etc/asterisk/mail_fax.pl ${CALLERIDNUM}
${CALLERIDNAME} ${FAXNAME})
exten = s,107,Hangup
And the result is only the first test getting executed.
Thanks,
Jeb Campbell
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James,
Thanks so much for taking the taking the time to help me figure this
out and learn something.
It is a Definity, but I'm not sure about the card -- just a basic t1
card is all I know (on Monday I could get more info).
Is there a command to find out which card is installed? or if that is
-- I just don't know what commands can troubleshoot the
connection.
Config:
zaptel.conf (relevant section)
span=1,1,0,esf,b8zs
fxsks=1-24
zapata.conf
[channels]
context = demo
switchtype = national
signalling = fxs_ks
group = 1
channel = 1-24
Thanks for any time and help
Jeb Campbell
[EMAIL
was no help).
If you are positive those directions are correct I will try again -- I
ask because on irc
#asterisk, some people thought those were wrong (like to use esf
framing).
Thanks for the response,
Jeb Campbell
[EMAIL PROTECTED]
Cell: 865-385-1437
On Mar 20, 2004, at 8:30 PM, James Coberly
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