Displaying 20 results from an estimated 43 matches for "yliu11".
2007 Jan 14
2
To 1.4 or not
I don't have a particular reason to upgrade, but I'm installing a new box,
so I have the opportunity to go 1.4. On the other hand, I'm not familiar
with 1.4, and relatively new to Asterisk. So instead of trying to keep up
with two different versions, I want to tie my handful of boxes to one,
before any of them grow too complex.
Is there a document about the main motivations to
2007 Feb 08
3
Asterisk and 802.11g
I'm greatly surprised when testing an Asterisk box with 802.11g. Here's the
topology:
VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension
|
FXO ___ PSTN extension
When I call a VoIP extension on that box (from a VoIP extension), voice is
good. But when this box tries to bridge the call with a
2006 Dec 15
1
fxotune unable to set impedence
My SM56 (Motorola X100P clone) has echo as hight as 38%, according to
fxotune -d. But when trying to take action, it fxotune simply says it
can't.
./fxotune -i3 -vvvv
Running with parameters:
doset=0
docalibrate=1
dodump=0
startdev=1
stopdev=252
calibtype=2
waveformtype=-1
delaytosilence=0
silencegoodfor=18
2007 Jan 03
3
voice fax modem and asterisk
Hi I have been asked to ind out if there is a way to use asterisk to
answere a voice fax modem so it can provide an answering service and
record messages ?
--
Gregory Machin
gregory.machin@gmail.com
www.linuxpro.co.za
2007 Jan 26
4
Does X100P decode caller ID?
The SM56 MODEM manual says it does. But when used with zaptel 1.2.12,
nothing shows up.
Yuan Liu
2007 Feb 04
1
TDM400 stopped bridging outgoing FXO call
My TDM400/zaptel 1.2.10/Asterisk 1.2.13 suddenly stopped bridging outgoing
FXO calls. If I make a call (from an FXS channel) to a PSTN destination,
and the other side answers, Asterisk will show continued ringback on the FXS
channel, while the PSTN side hears silence. No error message appears.
If a call from PSTN terminates on the same FXO, Asterisk can still ring the
FXS channel, and when
2007 Feb 04
1
Continue line in config files?
Is there anything that allows a logical line to extend to the next physical
line? Printed files are so hard to read with blind line wraps - and my
printer doesn't even automatically wrap.
Yuan Liu
2007 Feb 11
2
Extensions in macro
Home someone can explain this: a Goto() command can walk within a macro, but
if a digit is dialed from within a macro, the call flows back to the context
that called the macro. Is there some way to "contain" the flow? Thanks.
Yuan Liu
2007 Feb 22
3
New tutorial: DTMF tone detection
Hello list,
I have prepared a small tutorial today that deals with how to avoid
Asterisk rebuilding DTMF tones when using it to connect industial
appliances that use DTMF. You can find it at:
http://astrecipes.net/index.php?n=248
I know it isn't everybody's piece of cake, but I thought somebody could be
interested as well :)
l.
--
Home of QueueMetrics -
2007 Mar 02
1
How to fail an AGI
I mean how do I set failure condition in AGI? My script exits with code 0
upon success, and non-zero when problems occur - the standard *nix way. But
Asterisk always report "AGI Script completed, returning 0", and AGISTATUS is
always SUCCESS.
Yuan Liu
2007 Mar 05
2
Read() status?
Does application Read() return a status? Console displays stuff, but show
application read doesn't mention any status variable.
Yuan Liu
2007 Mar 06
1
Compiling smsq in 1.2
How to compile smsq in 1.2? It is compile in 1.4 by default. It is
included in 1.2.13, but not compiled. Any rule or method to make it?
Yuan Liu
2006 Dec 13
2
TDM400P won't ring GM phone of mere 0.1B
This is rather bizarre: My TDM11 (one FXS) rings a $10 passive phone with REN of 1.0B, a cheap speaker phone of 0.3B, and a cordless phone with marked REN of 0.0B. But it couldn't properly ring this 27935GE3-B (FCC ID G9H2-7930) cordless phone rated at merely 0.1B. Rarely, the phone will crack out an occasional weak and abrupt beap, but never a normal ring. Otherwise Asterisk and TDM400P
2007 Jan 05
2
SIP/TCP?
I'm still learning some of the basics. Can someone explain in layman's
terms what's the difficulty for Asterisk to support SIP/TCP (and even
RTP/TCP)?
2006 Dec 14
2
Console latency
Another bizarry: If I run the Echo application from the console, I can hear
a very long delay (upward to 1,000 ms). I can run the same application from
a GrandStream phone (on the same LAN) and hear little delay. What could
possibly be wrong? If it were interrupt overload, I'd hear lots of cracks
in my echo, right? I'm not hearing that. Besides, a telephony card is not
involved.
2007 Feb 26
7
How to get values of local channels context
The variable ${CONTEXT} stores the value of the current context. However if we are in a macro that will be the name of the macro. How do I access the name of the local channel's context.
For example:
[macro-test]
exten => s,n,NoOp(Context ${CONTEXT})
CLI shows:
-- Executing NoOp("Local/2592@1100006-2000-e802,2", "Context macro-test") in new stack
I want to get
2007 Feb 05
4
Having Trouble With Wait Command in Callback Context
I am trying to get called back with a DISA dial tone when I call a trigger
number. I got it to work almost the way I want, this is the callback
context:
[callback]
exten=> 501,1,Congestion()
exten=> 501,2,Hangup()
exten =>h,1,System(cp /etc/asterisk/callback.info
/var/spool/asterisk/outgoing)
exten =>h,2,Hangup()
With the above, the call comes into the trigger number, then the call
2006 Dec 15
1
ztmonitor displays full bar when idle
Hardware is an SM56 card (X100P clone). When the line hangs up, ztmonitor
displays full bar (or whatever maximum allowed by rxgain) in RX. It only
drops zero when the line picks up (and remote was silent). Is this
something of concern? The zap channel seems to work despite echo.
Additionally, what's the objective of tuning with ztmonitor? I mean, what
would indicate an optimal level?
2006 Dec 31
1
X100P "rings" randomly when "phone" line makes call
Not sure if anyone experienced the same - or if anyone ever connected a POTS
phone to the "Phone" jack on an X100P card.
The POTS phone rings normally when the FXO receives a call. The POTS phone
can also make outgoing calls when FXO is not holding the line. This is
desired. But if a call connected to the POTS phone lasts longer than a
couple of minutes, Asterisk would receive
2007 Jan 07
2
"Reserved" extensions?
I'm creating extensions _*XX. But whenever I press *0 or *8, Asterisk
throws out congestion and hangs up. I set verbose to 6 and debug to 6, but
all Asterisk cares to display in console is
-- Starting simple switch on 'Zap/1-1'
-- Hungup 'Zap/1-1'
Are these combinations forbidden?
Yuan Liu