Neil Cherry [EMAIL PROTECTED]
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1448547 Jun 22 15:23 /usr/local/lib/libiodbc.so.2.1.9
Have you done an ldconfig (or reboot) since your last compile?
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cards are sharing interrupts with another device.
Check /proc/interrupts.
How does one prevent the interrupts from being shared?
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either to another IRQ.
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unix 2 [ ] DGRAM2915 3221/
unix 2 [ ] DGRAM2847 3175/klogd
On Thu, 24 Jun 2004 10:11:46 -0400, Neil Cherry [EMAIL PROTECTED] wrote:
Caleb Kow wrote:
Hello Everybody,
I am trying to configure Asterisk to listen
and
that a change in the config is needed to get it working with TCP/IP.
I'd investigate that as that's what it looks like. I hope this helps.
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looking for a man page basically.
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Kevin Walsh wrote:
Neil Cherry [EMAIL PROTECTED] wrote:
OK, this may seem to be an obvious question but where do I find
the reference docs? I'm getting this error message:
Timeout, but no rule 't' in context 'home'
about this line:
exten = 2201,1,Dial(${PHONES1},20,Ttm)
I know the problem
Chris A. Icide wrote:
On 08:10 AM 7/1/2004, Neil Cherry wrote:
The WiKi is your friend.
So far it hasn't been very friendly. I tried to to find a document
I printed out (it printed poorly). When I entered the document's
title it fail to list that link. I actually found it via google
(weird
description of it means, I'm sure
I've missed a few details but that close enough for government
work.
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this either).
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on Win9x, but I have not tested it and can make no guarantees.
Thanks, I've been having real trouble with those stupid DLLs. I can't
upgrade some of them no matter what I do (WIN2K)!
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Kewlstart (Default) (Slaves: 01)
1 channels configured.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
Earlier I didn't get the above error.
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Ilia Mirkin wrote:
You want to use the wcfxo module with the X100P. wcfxs is for the
TDM400P card.
On Wed, 2004-10-13 at 03:43, Neil Cherry wrote:
I'm in the process of setting up the X100P card and I am getting
continuous tone on the X100P but only if plugged into the POTS
line. Here is what I
Maybe I'm just doing this wrong. Is the FXO card (X100P) used to
connect to the telco pots line?
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enough of this I
think I'll get another card maybe the TDM400P and an FXO to start
with. ARGH!
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[EMAIL PROTECTED] wrote:
On Wed, 13 Oct 2004, Neil Cherry wrote:
OK, I found the problem! The card must be bad, maybe the pair
is reversed. When I power down the server (power off, ac cable
disconnected) I still get the tone! I've had enough of this I
think I'll get another card maybe the TDM400P
I really found it now! I rewire the jack for 2 wires and I no
longer get the tone problem. I think it was crossed wires.
Now onto checking out Asterisk ...
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Jason Becker wrote:
Details of the project can be found here:
http://amp.voxbox.ca
One word of warning backup your settings (/etc/asterisk/*) before
installing it will overwrite your files.
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I just setup * to work with FWD and I'm now seeing these error messages:
IAX Packet 31216 from circuit ids 212-1conflicts with earlier call with
circuit ids 1-124
What can be causing this?
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to remove the
/var/lib/init.d/started/asterisk link otherwise Gentoo won't
let you restart the asterisk script (ARGH!).
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addresses).
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and it shows up in users and peers (but
not in registry). Where do I start with debugging?
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Author of: Linux Smart
Does the ${BLACKLIST()} function allow for values other than 1 to be
returned and if so how can I use that is the AEL? Can I use the
function in a switch statement?
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other services to
their appropriate servers. I have no mail box on the one account
(the one I'm testing to). I've followed the FWD instructions but
I've had no luck.
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do some more work and I'll post a new message under the proper
heading for that problem.
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pointers?
I've done a sip debug and all I'm seeing for the BT100 - Asterisk
is Asterisk asking the BT100 for it's option (102 Options) and
the BT100 not replying.
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this on the list so I thought I'd add it here.
Guess I should add it to the wiki (a little later).
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are your recommendations for a slightly fancy home phone?
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that she wants this
really ugly phone to be put in the kitchen. Looks like a phone
with a crank on the side. I know better than argue so I'll be
getting one of the Sipura gateway devices.
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: asterisk: command not found
Any ideas?
Login in as root, type in:
type asterisk
I get /usr/sbin/asterisk
Change crontab (crontab -e) to use the full path.
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.
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uucp ;
option tftp-server-name192.168.2.1;
always-reply-rfc1048 true ;
}
Here's mine, works under Linux.
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Il Neofita wrote:
I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however,
if I call the traffic still go throw the asterisk. How come?
Are you using the same codecs on the SPA3000 and the xlite? If no
then there's your reason.
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that in case
someone accidentally picks up my extra test phones.
BTW, this is for home use.
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= _1.,102,Playback(allison7/pls-try-call-later)
exten = _1.,103,Playback(allison7/all-circuits-busy-now)
exten = _1.,104,Macro(hangupcall)
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found this info).
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). This may be a good way to go and the cost should be less than
$100 (US). You'll still need the DSL modem but I don't like all in
one devices at that level.
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Does someone explain the 's' extension? In the Wiki it says it's
the catch all extension. In the Asterisk 1.2-rc1 it say it isn't
but doesn't say anything more. Needless to say I'm confused.
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Eric ManxPower Wieling wrote:
Neil Cherry wrote:
Does someone explain the 's' extension? In the Wiki it says it's
the catch all extension. In the Asterisk 1.2-rc1 it say it isn't
but doesn't say anything more. Needless to say I'm confused.
When a call comes into Asterisk (PSTN, VoIP, etc
Eric ManxPower Wieling wrote:
Neil Cherry wrote:
Funny thing is I can get Asterisk to use the 's' extension as a
catch all (I use the include = xcontext command). But I needed
to describe it properly for chapter in a book I'm writing. Man
I hope I get this stuff right!
BTW, I'm using SIP
was so bad at one point I was hearing multiple echoes.
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==
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I realized it was a joke.
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255.255.255.255 5061OK (15 ms)
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D 255.255.255.255 5060OK (16 ms)
2201/2201 (Unspecified)D 255.255.255.255 0 UNKNOWN
pstn/pstn 192.168.24.197 D 255.255.255.255 5061OK (16 ms)
3 sip peers [2 online , 1 offline]
Dang Grandstream!
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Neil Cherry wrote:
Humberto Aicardi wrote:
Neil,
When you use the Dial command you must specify the device to use
for dialing, so you cannot use Dial(2201,20) you must use
Dial(SIP/2201,20) which informs to use the the SIP device 2201.
Ah, thanks! That worked with a second extension I
but I think that's a problem with my
wireless access point.
How did you get access to the web config? What user and is it
the default password/access code?
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Paul Hayes wrote:
Neil Cherry wrote:
[snip]
How did you get access to the web config? What user and is it
the default password/access code?
type it's IP address into a web browser. Username: admin, password: psw
is the default.
Thanks, I was missing the psw.
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to POTS line (~2 cable miles
from the CO). I'm pretty sure my problems are software echo
cancelers doing battle with each other but I have no proof.
BTW, how do you turn off the software echo canceling in the SPA?
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://publibfp.boulder.ibm.com/epubs/pdf/wvrpad04.pdf
But it has been remove from the IBM site.
If anybody has a copy of this PDF file and can send me a copy it
would be really appreciated.
Thank,
Neil Cherry
to redirect one of the channels and
hold on to the other.
Thanks,
Neil Cherry
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On 2014-12-17 9:08 AM, Neil Cherry wrote:
Doe anybody know of a way to redirect both channels from a bridge to
different dial plan extensions from the using the AMI.
Currently, as soon as I redirect one of the channels the other appears
to be dropped and gets reorder tone (congestion, fast
On 2014-12-17 9:34 AM, Karsten Wemheuer wrote:
Hi Neil,
Am Mittwoch, den 17.12.2014, 09:08 -0500 schrieb Neil Cherry:
Doe anybody know of a way to redirect both channels from a bridge to
different dial plan extensions from the using the AMI.
Currently, as soon as I redirect one
it?
Is this something that changed (intentionally or accidentally) from
11.14 to 11.19 or I am missing something?
Thanks in advance,
Neil Cherry
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