Hello, I'm getting interested to purchase an item
that may solve my analog FXO/PSTN interfacing problems, as my national telco
doesn't give CPC or Battery Reversal (and is not likely to, ever).
Has anybody had experience with http://www.sandman.com/pdf/Page94.pdf,
CPC Generator from Sandman
hello, using Asterisk, is there any clever way to
provide answer supervision based upon the received audio only from the FXO
interface (from a public PSTN switch that does not have battery reversal, or
CPC).
I would like to use something like:
[toCALLOUT;script to call a particular number
Hello, I thought that my Digium TDM400P would be the right hardware to
support the zaptel timer, and put the following IAX.CONF entry to test,
(trunk=yes) in the example below
[VHAX]
type=peer
auth=md5
username=whoknows
jitterbuffer=yes
;trunk=yes
secret=terriblesecret
host=4.5.6.7
qualify=1200
hi,
i am trying to write a new tone/zone data set for
my home country. I edited zonedata.c, in ZAPTEL source code -- but then I
realised that I don't know exactly how to enable Asterisk to determine by
listening to the FXO interface connected to the PSTN what the progress of the
call is
hello, with reference to my earlier question
(sorry, I got confused), how does one use the "progzone" statement in
ZAPATA.CONF file ? How does on ensure that the 'progzone' command is parsed and
can be set to a NEW country definition, say, "tbd". Where does "tbd" have to be
defined ? Zaptel
Hello, anytime I make an IAX2 call to another peer,
I am collecting CDR records which are divided into small files, one for each
accountholder customer that makes the calls.
I have records of this nature:
""123456","1234567890","IAX2/[EMAIL PROTECTED]/5","2004-12-30
22:17:07","2004-12-30
Hello,
I would like to parse inbound Asterisk IAX2 7-digit numbers in the form of
123-4567 and strip out the first four digits, and then dial whatever number
digits remain. If I only have three digits (000-999) and have a mix of
channels (ZAP, SIP, IAX2) could someone please point out how I can
hi, I received this e-mail which contains a ballad, at first I thought it
was junk mail, but then I read through it, for the EE members of this list,
it may be quite humorous.
I don't know if the ballad is original, but at least it's the XMAS season,
so it's something to lighten up your day, eh?
Hello, is there a full guide to what kewlstart is supposed to do with FXO or
FXS lines ? is it only applicable to one of the interfaces FXO -or- FXS but
not both ? I asked earlier if FXS lines can be made to reverse polarity, and
someone else pointed out that the chipset on the FXS ports seems to
Ref: Message: 10
Date: Mon, 13 Dec 2004 08:31:04 -0700
From: Damon Estep [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Pitching Asterisk
http://www.millenigence.com/articles/asterisk-non-technical-review.pdf
Be careful about the last few paragraphs of this PDF, if possible, before
this paper is
[resend in plain text format]
On Tue, 2004-12-07 at 09:34 -0600, asterisk-users-
[EMAIL PROTECTED] wrote:
I've been struggling with a test * install for a couple months now in
a
small office and am just about ready to give up on it. It's not that
the
system itself is a problem. I've
You need to configure your legacy PBX to send a flash through from your
legacy phone to the legacy FXS port that is hooked to a * FXO port. This
is not something you can do from the * side.
*0 is asterisk's way of doing this. You need to find out if your legacy
PBX has a similar method. It
[this message is a resend, I apologise for duplicates, but I think the
original mail did not make it.]
12/3/2004
hi,
I followed the instructions faithfully in
http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3 and the directions were
very understandable and almost painless. I did miss the
hello, using a legacy PBX to access a Asterisk Zap channel (Legacy PBX
FXS -- FXO application Asterisk/TDM400P) I want to be able to flash the
asterisk pbx. However by pressing the FLASH button on the extension
connected to the Legacy PBX gets me the flash features on the Legacy PBX,
not on the
Hello, I want to use Asterisk PBX in front of my old, legacy PBX. The legacy
PBX can be outfitted with caller-ID and is already able to handle Calling
Party Control Signal Detection (this is a Panasonic KX-TD1232 Super Hybrid
PBX.
My question is how would one enable Asterisk to control the
On Tue, 2004-12-07 at 09:34 -0600, asterisk-users-
[EMAIL PROTECTED] wrote:
I've been struggling with a test * install for a couple months now in
a
small office and am just about ready to give up on it. It's not that
the
system itself is a problem. I've got everything (attendant,
Hello, I have found a bug, I think in the way TDM400P cards handle FXO
interface disconnect/re-connect problems. Normally I do keep all the wires
connected from my CO / PABX quite securely, but I had a need to re-route the
cable from one side of the desk to another, and I simply disconnected the
hi,
I followed the instructions faithfully in
http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3 and the directions were
very understandable and almost painless. I did miss the documentation, as
make progdocs would not work since the package doxygen was not included
in that WIKI guide, and
hi, we have just received our first shipment of
digium cards, FXO + FXS combinations, and collected all the hardware for our
custom clone server which will house our test-bed for asterisk.
I'm based in Dhaka, Bangladesh so you will
understand we may not always be able to get all the
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