Displaying 20 results from an estimated 1000 matches similar to: "Zapateller"
2004 Apr 12
3
Zapateller issues
Hi All,
In theory if I do this;
exten => s,1,Zapateller(nocallerid)
exten => s,2,Privacymanager
exten => s,3,Dial(a bunch of SIP extensions)
My callers should only hear the anti-telemarketing tones if they call from
a line that has no caller*ID and then get offered an opportunity to enter
it, right?
What I'm finding is that in the event of no CID the caller gets dumped
into the
2004 Oct 08
1
Zapateller Answering?
Been tinkering and found Zapateller appears to be answering when I didn't
expect it to. I have something like so:
[incoming]
exten => s,1,NoOp ;Zapateller(answer|nocallerid)
exten => s,2,PrivacyManager
exten => s,3,Dial(${RING},20)
...
I have a 1x1 analog * installation with a couple IP phones too. I've got
the FXO interface connected to the home phone line. When I get an
2004 Sep 15
2
Results of 13 month study on reducing telemarketing calls
Hello--
I've been playing with the privacy options on my home/home-office system
since August last year, and have some results, gleaned from my CDR
records, which over the last 13 months, number a total of 8672, which
includes incoming, as well as outgoing calls.
Before I start spitting out numbers, let me note that with the current
setup, I haven't had to tell a single telemarketer
2007 May 13
1
Zapateller and IAX2
Hi,
I have been using Zapateller with a TDM400 no problems at all, but
recently I have ported our BT number to a VoIP provider, and have a
strange problem. When I phone our number I first get the BT
unavailable three tone sound, and then it actually connects the call
via IAX2.
So, I disabled zapateller in the dialplan and tried again. Would you
believe it worked fine.
Has anybody else come
2003 Oct 24
1
Questions about Zapateller and Privacy Manager
Hey all...I'm just getting my * setup and right now all I have is an FXO
but no FXS. I wan't to get rid of telemarketers by having * pick up the
phone if there is no CID present, give the caller the Zapateller tones
and then ask the user to input their phone number via Privacy Manager
(yes I realize that this won't get us any where given that I can't
re-ring the phones without FXS
2004 Jan 21
3
Mailing List Lag
Has anyone from digium looked at why there is a 30 min to 3 hour lag on
messages on this list?
I.e looking at the last 50 messages I've received, the lag is about 90
minutes between the time sent and the time received.
Sometimes this drops to as little as 4 minutes.
Is this problem worse for me because my email address starts with "w" and my
copies of the emails get sent after
2007 Jun 12
4
GotoIf Dialplan inquiry
Hi all,
I have the following in my extensions.conf:
exten => s,4,GotoIf($["${CALLERID(number)}" = "8585979857" |
"8585970327"]?15:5)
The numbers listed above are known spammer numbers. However, when I call
from any other CALLERID, it still directs me to s,15 which is the
Hangup() application. Here are logs from the asterisk CLI:
-- Executing
2007 May 30
3
Dial plan inquiry using GotoIf()
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan. All I basically wish to do is block a
particular caller. Sounds easy enough, but my ternary operator/plan
currently is not properly being implemented. Can anyone spot where I'm
being a momo?
All extensions get forwarded to the following macro:
[macro-forward]
; arg1 = phone
2003 Nov 26
2
Issues with Privacy Manager and Zapateller
I am having issues with Privacy Manager and Zapateller.
If I set callerid="" on a sip user zapateller sends the tones
If I set callerid="Anonymous" <8475551212> zapateller doesn't send the
tones
If I call from a phone after dialing *67 zapateller doesn't send the
tones
In the last 2 cases, the display on the phone shows -Blocked Call-
PrivacyManager always gives
2004 May 24
1
Using Blacklist
I am attempting to write in incoming context for calls.
1. If the caller id is given and it is not black listed it will Playback a
greeting and then right the phone or go to voicemail under busy or
unavailable conditions
2. If no caller id is given, then Privacy Manager will ask for the number.
I am testing 6145551212 to see if the black list will work
3. If a caller id is given, and it is
2007 Mar 20
1
Zapateller not playing audio via SIP Trunk?
Hi All
I'm tracing a very strange problem which I could reproduce with different
versions up to 1.2.5 (sorry, didn't update to a newer one yet).
Scenario 1: Problem does not occure.
=============================
Sip Phone registered directly to the Asterisk.
exten => i,1,Zapateller()
exten => i,n,Playback(invalid,noanswer)
exten => i,n,Hangup()
Works like expected. I dial an
2003 Jun 12
2
Telemarketer GSM?
does anyone have a recorder GSM file that emulates the Telco's "if you are a telemarketer please hangup now" recording? I don't see one in the sounds dir. the ZapATEller works great for computerized callers but if a human hears this message asking them to go away they have to. Isn't that right?
Dave
2006 Apr 29
2
Codec G729 no longer works.
I upgraded my server from Fedora Core 4 to Fedora Core 5.
I was wondering if anybody else has run into the problem and know's the
fix?
I recompiled asterisk and if I don't have
the /usr/lib/asterisk/modules/codec_g729a.so
file in place it works.
I use or used to use the licensed G729 Codec from Digium.
This is the error message from asterisk -vvg:
[app_playback.so] => (Sound File
2005 May 28
1
3 goes and your out
Is it possible to give a caller three goes at an extension number then
hangup?
exten => s,1,Zapateller(answer|nocallerid)
exten => s,2,PrivacyManager
exten => s,3,Ringing(1)
exten => s,4,NoOp(${CALLERID})
exten => s,5,SetMusicOnHold(random)
exten => s,6,Background(silence/1)
exten => s,7,Background(thank-you-for-calling)
exten => s,8,Background(silence/1)
exten =>
2004 Jan 30
7
Calls dropping off
Hi,
I've got a fairly working Asterisk setup, with a few minor glitches, one of
which is very very irritating.
Sometimes, during a call, the remote end just drops off. We're using software
SIP phones (SJPhone) connecting to * then out through analogue lines with
X100P cards.
There is nothing in the logs and nothing on the console, the call just seems
to 'go away'!
Can anyone
2006 Dec 16
1
rxfax detection problems with multiple contexts
Hello,
I have a rather odd problem with Asterisk detecting faxes. I have two
POTS lines coming into the box (TDM400P). Line 1 is for voice, Line 2
is fof fax. When I set them up with channel => 1-2 in zapata.conf,
all is fine, but as soon as I have two channel => definitions,
Asterisk is unable to detect faxes. The fax line is not supposed to
ring local phones, so the most obvious
2004 Jan 19
2
Different Caller ID for each Zap Interface
Hi there,
I'm wondering if there is a way to assign a different Caller ID to each Zap
interface.
I have 3 Digium X100P cards, and I'm sure there must be some way of
configuring zapata.conf to allow each line to identify itself with a
different Caller ID string.
Many thanks,
Steve
--
Steve Foy | http://www.unite.net
UNITE Solutions | Tel: 028 9077 7338
2005 Jun 16
1
faxdetect config issues
My Asterisk fax detection used to work, but no longer does.
OK. So, here's the deal:
1. It appears that the "faxdetect" command cannot be applied
channel-by-channel in zapata.conf anymore, as Asterisk appears to the
last "faxdetect=" command to ALL channels.
2. My stations are detected and sent to the proper extension; i.e., when
I send a fax from one zap extension
2004 Jan 21
3
Making a call with sample.call
Hi there, I'm having some trouble with getting Asterisk to make a call, I
think it should be quite easy, but anyway...
Using the following file contents:
##
Channel: Zap/3/<TEL NUMBER HERE>
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: phones
Extension: 502
Priority: 1
##
Extension 502 is simply one that plays a sound back. When I dump this file
into
2003 Aug 05
3
Newbie just starting out with *
Hey all...I'm brand new to * and I want to convert my home into a pbx
type setup. I've figured out that I want a Wildcard X100P to bring my
single POTS CO into my Linux box. My problem is that I'm sure sure what
I need to do to get my analog phones connected up into a structured
phone system. It *looks* like I can go the route of the Cisco Analog ->
VOIP for about $100 on ebay.