http://www.elna-america.com/tech_al_reliability.php
Capacitors are one of the components on that motherboard that have a
finite life span. Other components are more or less tolerant of these
changes over time. Eventually the caps WILL fail...this could be 5
years or 25, but it WILL happen with
Hey FolksI am on the 1.2 branch with the latest from Subversion.
I've been having a rough go for the last several months integrating
asterisk with out Altigen system.
I can get calls inward just fine. I have zero missed interrupts on the
digium 110p card. I have zero frame slips according
We have an in house pbx that rings through to asterisk with auto
attendant asking for a password. It goes much too quickly to be
comfortable so I had to delay it 2 seconds before starting the auto
attendant. Folks like to dial their speaker phone then lift the
handset. During the phone base to
Just throwing out this question. integrating with
Altiware server. PRI appears to be okay. It keeps trying to move my
call to a different channel...usually channel 1. This is the deal
here:
Moving call from channel 23 to channel
1
Then the following errors after no audio then hanging
up
I could only get *ANI*DNIS* working one way and that was setting my
signalling type on the Asterisk side to 'featd' The Definity won't send
*ANI*DNIS* information back to the asterisk as far as I can tell.
Other than that, I've been running it with wink/wink EM for a while
now. TN464 circuit
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CVS Head No ringing on calling end?
Matt Loretitsch wrote:
Tie line is type em_w (old school stuff) to TE110P. Phone rings on
the asterisk side, but the calling party does not hear the ring
through sound
Tie line is type em_w (old school stuff) to TE110P. Phone rings on the
asterisk side, but the calling party does not hear the ring through
sound. If I pick it up within the first two rings it goes through and I
can talk otherwise our old switch drops the call.
Anyhow...here is my config if
Same method not allowed here on the CLI. Call from outside continues to
ring my internal polycom ip501 even after hanging up incoming call. CVS
from this morning. No solution yet. Hopefully tomorrow things will be
better.
-Matt
-Original Message-
From: [EMAIL PROTECTED]
I wish someone would just post a sample extensions.conf so I could
FINALLY understand this. Could you post at least the hint portion of
yours? I have tried this repeatedly without success and am starting to
feel like a true idiot.
Is it something like this?
exten = 2352,hint,SIP/2352
That's
Title: Message
/etc/zapata.conf
span=1,1,0,esf,b8zs (adjust to suit your t1 framing and
encoding)em=1-24
/etc/asterisk/zaptel.conf
[channels]echocancel=yesechotraining=400context=defaultsignalling=featd
(this send *ani*dnis* to my PBX, otherwise use 'em_w' for this
Motorola bluetooth headset that came with a cell phone
D-link blue tooth usb dongle
Eyebeam or x-lite software. Doesn't answer, but does work for audio in
and out.
-Matt
-Original Message-
From: Laurent Lesage [mailto:[EMAIL PROTECTED]
Sent: Monday, May 23, 2005 3:25 PM
To: Asterisk
I believe *ANI*DNIS
That's how Asterisk sends it when I set my t1 line to featd.
In /etc/asterisk/zapata.conf
signalling=featd
not much to go on, but a little!
-Matt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: Thursday, May 12,
Title: Message
Sure!
exten=_82XXX,1,Dial(SIP/${EXTEN:1},15,tr)exten=_82XXX,2,Voicemail(u${EXTEN:1})exten=_82XXX,3,Hangupexten=_82XXX,102,Voicemail(b${EXTEN:1})
This
let's someone dial an 8 first to dial a 4 digit SIP extension. Normally
everything goes out through the T1 line to our Definity
Anyone using FOP or another solution to watch 200 + lines at once? I
don't think I can cram the buttons down small enough in FOP to do it!
Currently they are used to the Avaya Definity panel with the whole
decade setup. I may need to deploy over 200 phones and have them all
visible to the
I am using a Dell Optiplex G1. Unable to ascertain who made the motherboard
for them at that time. It doesn't look familiar internally. Maybe I'll switch
PCI ports and see what comes of it. Mine doesn't appear the share any
interrupts either. The zttest gives me 99.999% for a 5 minute
Looking for some help any way I can. I've been closely following
digium's troubleshooting steps and seem to be okay there. I am
connecting, via PRI, to a Definity system. When I release the board on
the Definity side I get this in Asterisk:
*CLI Apr 7 10:17:23 NOTICE[13099]: chan_zap.c:7395
16 matches
Mail list logo
