Yes, the dialplan field in the PAP (not asterisk's dialplan) was the
problem. The dialplan used to have *xx in it (as well as lots of other
stuff which we left alone), we changed that to *xxx (leaving the double
*'s in all of the vertical service activation codes) and it now works.
thanks
I wrote:
In the PAP2's setup there are all of these Vertical Service Activation
Codes that start with star and Outbound Call Codec Selection Codes,
also the setup menu is accessed by pressing star four times, could they
be intefering with dialing numbers that start with a star? And is there
We have some extensions in our dialplan that start with a star. We can
dial them from Zap phones and SIP phones, but not from phones connected
to a PAP2. After the user presses star follwed by two digits (our
extensions are dialed with star followed by three digits) he hears a
fast-busy that
I have a Linksys PAP2. Identical setups for the two channels in both
the unit and in Asterisk. In particular, both channels enable g729 and
set it as the preferred codec, and have disallow=all and allow=g729 in
sip.conf.
If we make a call on one channel, it works (and uses g729), but if we
Dan Elder wrote:
Hey all, I've been seeing this repeatedly and am wondering if anyone has a clue
what's causing it.. at least once a day I see two zap fxo channels being
bridged, and hanging..now, these two channels should never bridge, but they
keep doing it.. any leads on where to look for
I see these from time to time, I think it means that packets got lost,
or received out of sequence. It looks to me like asterisk manages to
deal with this, so unless your calls have also stopped working, I
wouldn't worry. (If we should be worrying, I expect someone will let us
know).
Leo Ann Boon wrote:
Warren Burstein wrote:
I've observed a situation on my production system, and have managed
to recreate it on my test system (both running 1.2.4). I pick up a
phone connected to a TDM400B's FXS line. I dial a number (in my
tests, it was another local phone
I've observed a situation on my production system, and have managed to
recreate it on my test system (both running 1.2.4). I pick up a phone
connected to a TDM400B's FXS line. I dial a number (in my tests, it was
another local phone, but in production it was an outside call), and that
call
it does rotate files on SIGFSZ,. it should rotate the csv file,
too, and any other files that are written to (maybe only of they are
larger than the file size limit)
Kevin P. Fleming wrote:
Warren Burstein wrote:
How about if it would set a global variable before each disk write so
Kevin P. Fleming wrote:
Dov Bigio wrote:
Any way, if any developers are reading this, I don't think that rotating
asterisk logs is the best way to handle this problem!
Maybe a more user-friendly message could be logged, infoming which file
reached the 2.0GB.
Unfortunately when we
app_txfax.so
cdr_addon_mysql.so
chan_modem_aopen.so
chan_modem_bestdata.so
chan_modem_i4l.so
chan_modem.so
format_mp3.so
res_config_mysql.so
Warren Burstein wrote:
Julian Lyndon-Smith wrote:
These modules are not part of the standard 1.2.3 release - did you
also install the 1.2.3 release of the asterisk
Julian Lyndon-Smith wrote:
These modules are not part of the standard 1.2.3 release - did you
also install the 1.2.3 release of the asterisk-addons package ?
The lastest asterisk-addons I found at
http://ftp.digium.com/pub/asterisk/ is 1.2.1. The only module I use is
cdr_addon_mysql.so. I've
I didn't find that exact message in the RFC's, but I did find something
similar in RFC 3407 (http://www.rfc-archive.org/getrfc.php?rfc=3407),
a=cdsc: 4 image udptl t38
Which means that the sender is capable of sending T.38 fax over UDP.
I wouldn't worry about it unless you were trying to
I thought I sent this earlier this week, but I didn't see it. If I
missed it, I apologize for the resend.
We are running asterisk 1.2.2 with a TDM04B connected to PSTN lines. On
incoming calls from cellphones located overseas, DTMF is not recognized
- we have many single-digit choices in
We have PSTN lines connected to FXO lines of a TDM400B. I just got a
complaint that overseas callers who are using cellphones sometimes find
that DTMF digits aren't working - they press digits and the menu goes on
as if they hadn't pressed anything. Since it sometimes works, and other
IVRs
I'm running asterisk 1.0.9 with TDM400B's for both internal and external
lines.
I put in the macro that dials outside lines an AbsoluteTimeout(36000),
never expecting it to happen. But it does, a few times a month.
I've noticed an odd thing, it seems that it usually happens twice in a
row
Simone Cittadini wrote:
Warren Burstein ha scritto:
What is frustrating is that the cdr file shows the dst as T rather
than as the phone number dialed. I realize that AbsoluteTimout
causes it to jump to the T extension, but it would help to know who
the user dialed (asking a week later
Tim Litwiller wrote:
Well, I'd like them to drop in my voicemail when done recording -
maybe in a separate recordings folder but I'd like to use the same
interface to play them back.
I would like that, too. Is anyone working on it? If not, I will put it
on my TODO list.
Recently I put callwaiting=yes in zapata.conf because customers want to
speak to the operator in person, not leave her a voicemail, when she's
busy with another caller. But now she can't transfer either of the
calls (which she can do when there's only a single call).
The operator has an
Arik Funke wrote:
can anybody tell me how to create an extension that starts with a *?
The expression matching works well if * is embedded in numbers but if
the extension starts with *, it is not executed but extension s
instead. Is there another way besides using a lot of if statements in
Joerg Wleklik wrote:
Hi Folks,
Does anybody have experiences with plugging 3 TDM400P cards in one PC??
I think about a Asterisk box handling 8 incoming analogue lines and providing
4 lines to an old analogue PBX.
I read a lot about trouble with the TDM400P cards so this idea seams to be not
C F wrote:
I wish you would know what you are talking about.
We've got four TDM400P cards in one PC running without any trouble,
since January.
Good for you, most people don't have it this way.
It's not a fair comparison between new cards that come from
Warren Burstein wrote:
I'm going to be hooking FXS lines on a TDM400 to a PBX which doesn't
drop line voltage at the end of a call, so I'm going to have to use
busy detection. A few questions -
The tones are taken from the tones specified by the zone in
zaptel.conf, right? Which tones cause
I've noticed that some callers listen to our main menu and don't press
any keys. I have it set up to restart the menu a few times and
eventually hang up. I'm wondering if these are wrong numbers (in that
case, why don't they hang up) or they really want to speak to someone
here but don't
We are using Bellsouth 8867 phones on our TDM400B FXS lines
(asterisk-1.0.3). It has a Voicemail light, which appears to be MWI
(according to the manual it works with voicemail from the telco that
sends a FSK signal). The dialtone stutters when a line has voicemail, so
I know that I have the
I'm not having any trouble with interrupts, but here's my
/proc/interrupts on Fedora Core 2 on a hyper-threading CPU and using the
SMP kernel (2.6.5-1.138). I don't think I need to worry about uhci_hcd,
nothing is plugged into USB, but libata is the disk driver. How do I
get libata and wctdm
Michael Welter wrote that I should be worried about the usb module.
Would rmmod uhci_hcd be enough, or should I disable it in the BIOS
like Shoval said?
Also, after the rmmod, I still have the conflict with libata on 169
CPU0 CPU1
0:73110067252568IO-APIC-edge
I moved my TDM400B cards (first two cards are 40's, third is a 31, last
is an 04) from one computer to another, copied all the config files, and
now the LED on the line 11 - third line of the third card doesn't go on
(it used to on the previous computer). I can get by telling * not to
use
Here's a strange one - when I run safe_asterisk on either of these
distros, words that are colored blue or violet (but not red) turn up in
Russian (and some other languages, I think). If I run asterisk with the
same arguments (-vvvg -c) as safe_asterisk does, from the console, it's
OK. If I
extensions.conf has
ignorepat = 9
exten = _9X.,1,Dial(Zap/G2/${EXTEN:1})
The first user to try it asked if instead of keeping the same dialtone
after pressing 9, if I could play a different dialtone. Can this be
done? I'm running asterisk 1.0.0 in case that matters.
Rich Adamson wrote:
How old of code are you looking at? The wcfxs driver was renamed to wctdm
some time ago. Current cvs doesn't include it.
I was using zaptel-1.0.0, but I took a look at 1.0.3 and it's still wcfxs
there. I'm about to bring this online, would rather stick with releases.
One Zap FXS channel has dialed to another. Zapata.conf has transfer = yes
and threewaycalling = yes. I flash on one of the phones, the other gets the
music on hold. If I hang up the flashed phone, it rings back and I am
reconnected to the other phone.
Is there some way (with flash, not with #)
I have more FXO ports on TDM400's than I have PSTN lines available for
testing. When all the lines were used up (the FXO ports are all in
zap group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial
succeeded even though there is neither line voltage nor dial tone.
Can at least
I came across the same problem today.
Phase one of our * project is to replace the PBX in one of our offices with
*, and one of the extensions will be sent over VOIP to service
representatives at a different location. But as a fallback, we want to dial
directly if VOIP doesn't work (maybe the
I have more FXO ports on TDM400's than I have PSTN lines available for
testing. When all the lines were used up (the FXO ports are all in zap
group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial succeeded
even though there is neither line voltage nor dial tone. Can at least the
lack
The FXO lines on my TDM400 are connected to PSTN lines in Israel. As far as
I know, the lines around here have disconnect supervision (I've seen some
other Israelis on this list, anyone know for sure?), because it's worked on
Dialogic cards, which reported hangup, not busy detect (while when I
Do you have threewaycalling and transfer set in your zapata.conf? Here's
mine (four TDM400's, seems to be working so far). I didn't do anything in
my extensions.conf for any of these features (what confused me at first is
the t and T options of the Dial application in extensions.conf are for
http://lists.digium.com/pipermail/asterisk-users/2004-September/061677.html
says If you don't have a soundcard then try just loading the sound module,
that might just be enough. I don't know what that means. What sound
module? In Asterisk? In Linux?
Yes, I have mpg123 0.59r
I
think Ive managed to figure out that there are two ways to transfer a Zap
call, using hookflash (defined in zapata.conf) or the # key (the t and T
options of the Dial command in the dialplan), but not why there are two ways to
do this, nor what the difference is between them. Is there
Is
the power connector on the TDM400P only needed for line and dial voltage, or do
you also need it if it has all FXO lines?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
When I first saw the priority numbers in extensions.conf, I thought BASIC,
if a number is missing, * will fall thru to the next number. I learned that
this is not so, if you have nothing between 1 and 3, you don't ever get to
3.
But I'm wondering what does happen? Hangup and wait for next
Ariel Batista wrote:
Warren Burstein wrote:
One more thing about prompts, it's better to say for sales press 5
than press 5 for sales, because by the time you hear sales you've
already forgotten what number it was.
If you add the sounds all you need is For Sales recorded the new sounds
I
thought I had posted this, but I didnt see it in the archives, so I
guess I hadnt.
Ive
got FXS lines going to a legacy IVR. When I Dial into one of these lines
and then hang up, FXS plays the Congestion tone until the IVR drops voltage. I
would like the IVR to hang up sooner. I could
One more thing about prompts, it's better to say for sales press 5 than
press 5 for sales, because by the time you hear sales you've already
forgotten what number it was.
So record for sales press and the digits (you could use the digits that
come with *, but a sentence in two voices sounds very
I connected two plain old telephones to FXS lines of a TDM400P (defined as
fxoks in zaptel.conf), one of them dials the other with Dial(Zap/2), I talk
to myself for a while, hang up either of the phones, and the phone that
remains off-hook gets the congestion tone until it goes on-hook (at least
I
have:
RedHat 9.0
TDM40B
asterisk-0.9.0 compiled from sources
zaptel-0.9.1 likewise
/etc/zaptel.conf contains
fxoks=1-4
loadzone = us
defaultzone=us
loaded
modules zaptel and wcfxs
/etc/askterisk/zapata.conf
contains
[channels]
language
= en
signalling
= fxo_ks
That did it. I thought Ringing() did that, but I guess it's just for when
you want to fake a ringing tone. I'll add a comment to
http://www.voip-info.org/wiki-Asterisk+cmd+Ringing.
thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bart Coppens
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