Dear
I am using an asterisk 1.2.7.1 , with postgres
and safe_Asterisk, for running, asterisk.
but there is a problem,
after 2-3 hours after restarting any things, top
shows me, that, two asterisk, are now running, and one
of them, gets 99.7 percent of cpu.
Do you have any idea?
Best
Mani
show us the output of ur top command
Pezhman Lali wrote:
Dear
I am using an asterisk 1.2.7.1 , with postgres
and safe_Asterisk, for running, asterisk.
but there is a problem,
after 2-3 hours after restarting any things, top
shows me, that, two asterisk, are now running, and one
of them, gets
thanks Benjamin
the folowing is the output of TOP.
Best
top - 08:23:09 up 15 days, 2:26, 2 users, load
average: 1.31, 1.29, 1.24
Tasks: 109 total, 2 running, 107 sleeping, 0
stopped, 0 zombie
Cpu(s): 97.0% us, 3.0% sy, 0.0% ni, 0.0% id, 0.0%
wa, 0.0% hi, 0.0% si
Mem:450456k
Am Dienstag, den 25.09.2007, 07:37 +0200 schrieb Guenther Sohler:
Hallo Group,
I have basically set up a small asterisk system,
which ahs 4 peers:
* registers at 2 Sipgates
* 2 hardware phones connected to it
Both Hardware phones can phone outwards(cheaper sipgate is selected with
Sorry for this. This is most likely a HOWTO or FAQ question, but
it's so much information and documentation to wade through so
I hope someone could take a minute to answer anyway.
If not, no worries. I'll get to it sooner or later :)
I'm trying to understand what Asterisk actually is and the
Hi,
I'm sending this mail here as I think it is of interest for at least
europeans amongst you. The event presented below is not
commercially-oriented, but really of interest for professional
Asterisk users.
in two weeks (910 october), the event Asterisk and VoIP in
enterprise will take place in
Turbo Fredriksson wrote:
Sorry for this. This is most likely a HOWTO or FAQ question, but
it's so much information and documentation to wade through so
I hope someone could take a minute to answer anyway.
If not, no worries. I'll get to it sooner or later :)
I'm trying to understand what
Hi,
On 9/25/07, Turbo Fredriksson [EMAIL PROTECTED] wrote:
Sorry for this. This is most likely a HOWTO or FAQ question, but
it's so much information and documentation to wade through so
I hope someone could take a minute to answer anyway.
If not, no worries. I'll get to it sooner or
Hi,
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a
business graded installation (with really traffic on not 3 calls a
day ;-) )
Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall)
Thanks!
Kind Regards,
Erik
Hi list,
My Asterisk config for outgoing calls is the following:
exten = s,1,Dial(SIP/[EMAIL PROTECTED],60,g)
exten = s,n,GotoIf($[\${ANSWEREDTIME}\ = \\]?pri:hang)
exten = s,n(pri),NoOp(Problems with voip provider trying PRI)
exten = s,n,Dial(Zap/g2/${MACRO_EXTEN},60,g)
exten = s,n(hang),HangUp
I have sent a few emails over the past couple of days that simply have
not arrived on the list (or so it seems).
Is anyone else encountering this ?
Julian
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth
Dear Anselm,
I am sorry about the big traffic in the newsgroup.
I tried to send my post to the newsgroup for 3 days now - once a day,
but it did not appear. Today I tried putting it in cc also with, then it
worked out ...
I will carefully read your answer.
thank you veryy much
Hi All,
I'm interested in how people are clustering Asterisk, if that's possible, or
how you might be achieving a redundant solution.
I've a single Asterisk server driving the company. Its well backed-up, and
I've a cloned machine that (in theory) with a DNS change could take over
operations.
Marc Patino Gómez wrote:
in most cases it works well but, if my internet connection is down
Asterisk tries to Dial voipprovider, but it can't resolve the dns name,
so it waits 60 seconds to jump to the following priority...
Any ideas to solve this problem? I can't use the IP of the
Yes, I have read some articles about it.
But I would like to try something similar to
http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/1939626.html
Any experience in this?
From: Jon Schøpzinsky [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Yes, I have read some articles about it.
But I would like to try something similar to
http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/1939626.html
Any experience in this?
From: Jon Schøpzinsky [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Yes for me.
Carlos
--
Julian Lyndon-Smith wrote:
I have sent a few emails over the past couple of days that simply have
not arrived on the list (or so it seems).
Is anyone else encountering this ?
Julian
___
Sign up now for AstriCon 2007!
Hi all,
I have installed X-lite client on a windowsXP
machine and asterisk on an enterprise linux m/c.
The client is sending a registration message to asterisk
server. It is able to process the message and sends 200 OK
back. But later it says Scheduling destruction of sip
dialog . Then
On 22:43, Tue 25 Sep 07, Carlos Hernandez wrote:
Yes for me.
Carlos
--
Julian Lyndon-Smith wrote:
I have sent a few emails over the past couple of days that simply have
not arrived on the list (or so it seems).
Is anyone else encountering this ?
Julian
I have similar
Hi Adam,
thanks for your quick answer, I try your tip but the problem persist,
so... It seems not to be a dns problem
Asterisk executes the Dial command and it tries to reach the VoIP
provider until timeout, in * console appears:
Called [EMAIL PROTECTED]
Anybody knows howto make dial
Hola Jonathan
Te cuento un pokillo lo q intento hacer por si me puedes orientar en algo o
de algun sitio donde pueda mirar
Existe una especificación de Microsoft de lo que llaman
Dual-Forking, que básicamente consiste en poder usar tanto el teléfono como
el
propio PC como dispositivo de
Qualify=yes?
Thanks,
Steve
Marc Patino Gómez wrote:
Hi Adam,
thanks for your quick answer, I try your tip but the problem persist,
so... It seems not to be a dns problem
Asterisk executes the Dial command and it tries to reach the VoIP
provider until timeout, in * console appears:
me too :)
Original-Nachricht
Datum: Tue, 25 Sep 2007 12:57:25 +0200
Von: Michiel van Baak [EMAIL PROTECTED]
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Anyone else having problems with the list
On 22:43, Tue 25 Sep 07, Carlos Hernandez wrote:
Yes
Am Dienstag, den 25.09.2007, 11:01 + schrieb dadsadsadf dsadasdsa:
Hola Jonathan
Te cuento un pokillo lo q intento hacer por si me puedes orientar en algo o
de algun sitio donde pueda mirar
Existe una especificación de Microsoft de lo que llaman
Dual-Forking, que básicamente consiste
Doug wrote:
I was progressively upgrading this phone from 3.1.2
to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but
when I went to 4.0.0 (Even adding the more specific
2345-11500-040.bootrom.ld), it won't run, and
just keeps rebooting.
Now I've got a really nice doorstop unless someone
Hi List;
If I am configuring Diguim Analoge card, then I need
to run #modprobe wctdm, but the question why I need to
run #modprobe zaptel also?
What #modprobe zaptel does a things that #modprobe
wctdm does not do?
Any help?
Regards
Bilal
On Tue, 2007-09-25 at 10:14 +0100, Julian Lyndon-Smith wrote:
I have sent a few emails over the past couple of days that simply have
not arrived on the list (or so it seems).
I'll take a look at this again... I thought we had most of the problems
with the mailing lists fixed, but we seem to be
On Wed, 2007-09-26 at 03:49 +0630, Treesa Fairy Joseph wrote:
Hi all,
I have installed X-lite client on a windowsXP
machine and asterisk on an enterprise linux m/c.
The client is sending a registration message to asterisk
server. It is able to process the message and sends 200 OK
back.
On Tue, Sep 25, 2007 at 05:55:13AM -0700, bilal ghayyad wrote:
Hi List;
If I am configuring Diguim Analoge card, then I need
to run #modprobe wctdm, but the question why I need to
run #modprobe zaptel also?
No.
What #modprobe zaptel does a things that #modprobe
wctdm does not do?
On Tue, 2007-09-25 at 05:55 -0700, bilal ghayyad wrote:
If I am configuring Diguim Analoge card, then I need
to run #modprobe wctdm, but the question why I need to
run #modprobe zaptel also?
The wctdm kernel module depends on the zaptel module, so the zaptel
module will get automatically
In article [EMAIL PROTECTED],
Barton Fisher [EMAIL PROTECTED] wrote:
We have a Te410P with 3 Telco T1's (D4 SF ) with DID's (non-PRI). ANI
DNIS is received in-band DTMF in a format such as *7145551212*8002*
What happens when there are 30 or more calls, asterisk might see is DNIS =
802
Adrian Marsh wrote:
I'm interested in how people are clustering Asterisk, if that's
possible, or how you might be achieving a redundant solution.
I've a single Asterisk server driving the company. Its well
backed-up, and I've a cloned machine that (in theory) with a DNS
change could take
Hi All,
I'm compiling zaptel. Did the usual ./configure, then
make. Compile breaks saying:
/usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown
field âownerâ specified in initializer
/usr/src/zaptel-1.4.5.1/wcusb.c:1451: warning:
initialization from incompatible
Nevermind, I found the answer on the wiki:
Want to run an external program whenever a caller leaves a voice mail
message for a user? This is where the externnotify command comes in
handy. Externnotify takes a string value which is the command line
you want to execute when the caller
On Tue, 2007-09-25 at 15:59 +0200, Per Jessen wrote:
I haven't looked into it in any detail, but how about the standard Linux
HA solution with a heartbeat monitor, a shared file-system and IP
take-over?
It's been my experience that this usually works fairly well for
stateless protocols like
Hi all,
does anybody know any way that when it run reload app_queue in the
asterisk cli it don't lose the informations from show queue (queue
name) ?
I'm passing for this trouble, because I need this informations
(http://www.voip-info.org/wiki/index.php?page=asterisk+cli+command+show+queue)
Is there a way to tell asterisk, via a sip.conf peer, what IP address to send a
packet out of?
I've got multiple NICs in my box, each with it's own public IP. I need the SIP
messages to originate from any one of the IPs depending on which number was
originally called(and therefore where the
On Tue, Sep 25, 2007 at 03:22:01PM +0100, Jeng Yu wrote:
Hi All,
I'm compiling zaptel. Did the usual ./configure, then
make. Compile breaks saying:
/usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown
field âownerâ specified in initializer
Hi all,
I hope that I'm not breaking protocol too much by posting a message in this
group about a problem that I'm having with an Asterisk Business Edition
installation, but the reason that I'm posting here is because the problem
that I'm having isn't really with the Business Edition, it is
Sure,
Heres a basic overview:
- All IP (no local E1/T1 connections).
- 2Mb Fiber internet pipe backed up by a DSL backup.
- Single Asterisk server (with a backup clone on standby). Config
currently backed up to SVN and copied off by tarball by webmin to a
separate network.
- Both IAX and SIP
Adrian Marsh wrote:
I'm interested in how people are clustering Asterisk, if that's possible,
or how you might be achieving a redundant solution.
I've a single Asterisk server driving the company. Its well backed-up, and
I've a cloned machine that (in theory) with a DNS change could take
On Tue, 2007-09-25 at 10:57 -0400, Larry Costigan wrote:
Hi all,
I hope that I'm not breaking protocol too much by posting a message in
this group about a problem that I'm having with an Asterisk Business
Edition installation, but the reason that I'm posting here is
because the problem
Sorry, I should have mentioned it in my mail.
uname -r gives:
2.6.15-1.2054_FC5smp
Thanks,
Jeng
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Sep 25, 2007 at 03:22:01PM +0100, Jeng Yu
wrote:
Hi All,
I'm compiling zaptel. Did the usual ./configure,
then
make. Compile breaks
Everton Goularth wrote:
does anybody know any way that when it run reload app_queue in the
asterisk cli it don't lose the informations from show queue (queue
name) ?
I'm passing for this trouble, because I need this informations
Also, /usr/bin/gcc --version gives:
gcc (GCC) 4.1.0 20060304 (Red Hat 4.1.0-3)
Copyright (C) 2006 Free Software Foundation, Inc.
Also, /usr/bin/make --version gives:
GNU Make 3.80
Copyright (C) 2002 Free Software Foundation, Inc.
Thanks,
Jeng
--- Jeng Yu [EMAIL PROTECTED] wrote:
Hi All,
My problem:
Sometimes the sound seems to cut on calls in progress. We (on our local
SIP phones, Thomson ST2030's) can't hear the remote caller. The caller
may hear some kind of horrid sklurk and then it goes dead for them too.
Our Asterisk is connected to the France Telecom network by an Eicon
Upgrade your kernel.
Run:
# uname -r
if you do not see smp in the kernel version
Run:
# yum update kernel kernel-devel
If you do see smp
Run:
# yum update kernel-smp kernel-smp-devel
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
On Sep 25, 2007, at 10:53 AM, Tzafrir Cohen wrote:
On
Adrian Marsh wrote:
so maybe it's a case of looking at
Linux-HA.
If I remember this correctly a normal ping is all Linux HA can
do. It does not check whether Asterisk or other services are
alive and respond to queries.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied
On 9/25/07, Everton Goularth [EMAIL PROTECTED] wrote:
does anybody know any way that when it run reload app_queue in the
asterisk cli it don't lose the informations from show queue (queue
name) ?
A 'keepstats' option has been added to -trunk, and will show up when 1.6 is
released. Until
On Tue, 2007-09-25 at 18:01 +0200, Philipp Kempgen wrote:
Adrian Marsh wrote:
so maybe it's a case of looking at
Linux-HA.
If I remember this correctly a normal ping is all Linux HA can
do. It does not check whether Asterisk or other services are
alive and respond to queries.
Have
On Tuesday 25 September 2007 09:22:01 Jeng Yu wrote:
/usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown
field âownerâ specified in initializer
Type 'make menuselect', deselect wcusb, then left-arrow
out to the top, hit 's' for save, then 'make' again.
--
Tilghman
Dave Walker wrote:
On Tue, 2007-09-25 at 18:01 +0200, Philipp Kempgen wrote:
Adrian Marsh wrote:
so maybe it's a case of looking at
Linux-HA.
If I remember this correctly a normal ping is all Linux HA can
do. It does not check whether Asterisk or other services are
alive and respond to
It's nice to see Asterisk redundancy being discussed. A year and half ago, when
I posed the question of Asterisk redundancy, I was looked at like I was from
outer space.
- Original Message
From: Jared Smith [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Tue, 2007-09-25 at 10:59 +0200, Erik Wartusch wrote:
Hi,
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a
business graded installation (with really traffic on not 3 calls a
day ;-) )
Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall)
Nagios that's not redundancy.
- Original Message
From: Dave Walker [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, September 25, 2007 9:09:46 AM
Subject: Re: [asterisk-users] Asterisk Redundancy
On Tue,
I have a client using the Grandstream phones (not sure which model but it looks
fairly low-end) and they're lukewarm on them. The display doesn't tilt up for
easy viewing and the sound quality on the speaker phone leaves something to be
desired apparently.
But as basic, inexpensive, Asterisk
On Tue, 2007-09-25 at 12:10 -0500, Douglas Garstang wrote:
Nagios that's not redundancy.
And a brick isn't a house.
Clearly you know what Nagios is; and it's support for event-handlers.
If you had taken a moment to think, then you would know Nagios can form
part of a redundancy system.
I am having a similar issue with 4.0.0. Mine is that it doesn't get any
DHCP address (gets stuck waiting for an address).
I fixed it by going back one to the previous bootrom version, worked like a
charm.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Philipp Kempgen wrote:
Adrian Marsh wrote:
so maybe it's a case of looking at
Linux-HA.
If I remember this correctly a normal ping is all Linux HA can
do. It does not check whether Asterisk or other services are
alive and respond to queries.
I think the basic Linux-HA setup works with
Philipp Kempgen wrote:
I don't want to quote my text as not to spam the list (although
it's all GPL). There's a nice countdown at
http://www.amooma.de/gemeinschaft/
Very nice. I'll have to come back and take a closer look sometime.
/Per Jessen, Zürich
--
http://www.spamchek.com/ - your
On Tue, 25 Sep 2007, Ben Schorr wrote:
I have a client using the Grandstream phones (not sure which model but
it looks fairly low-end) and they're lukewarm on them. The display
doesn't tilt up for easy viewing and the sound quality on the speaker
phone leaves something to be desired
Greetings list,
I need to set up a point to point SIP connection between two devices without
either of them registering with a registrar/proxy/etc. at all. The devices I've
tested so far all seem to insist on having a registration before they'll make
or take calls.
One of the devices needs to
Run it with a stock OpenSER installation that will accept registrations
and acknowledge them with a 200 OK, but not actually do anything with
them.
On Tue, 25 Sep 2007, Chris Bagnall wrote:
Greetings list,
I need to set up a point to point SIP connection between two devices without
either
JM == Jeremy Mann [EMAIL PROTECTED] writes:
I would have answered, but I was prohibited from quoting properly:
JM If you are the intended recipient, further disclosures are
JM prohibited without proper authorization.
/Benny
___
Sign up now for
Why did you waste time with this reply? You do realize some users don't have
control over their Exchange servers, and asinine footers are placed into an
email without their intervention or control right?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of
since asterisk is only using operating system's routing ability , you can
always set static routes using route command in linux .
On 26/09/2007, Jeremy Mann [EMAIL PROTECTED] wrote:
Why did you waste time with this reply? You do realize some users don't
have control over their Exchange
Bilal,
The '#' symbol is part of a root prompt, not the command. In fact, if you
run a command in this way, it will not work because the shell will
perceive you as trying to enter a comment, as one would do in a
shellscript.
The Zapata modules have a series of interdependencies based on the
And if the Sip provider is sending data from 1 or two fixed hosts?
For instance, they send DID1 to IP A.B.C.D from 1.1.1.1
They send DID2 to IP E.F.G.H from 1.1.1.1
How do you differentiate? Would fromhost= work?
This e-mail, facsimile, or letter and any files
A friend of mine just sent me this email - he is looking for an IAX
inbound service in Holland - any thoughts?
Voip info only has Nadiz which seems to be SIP only.
Hi Dean,
I need a Dutch number with IAX support. Do you have any leads in that
direction? It's been difficult for me to
zoachien == zoachien [EMAIL PROTECTED] writes:
zoachien Turbo Fredriksson wrote:
How do I connect to a 'normal' (i.e. analog) telephone?
zoachien - you can take a voip provider and not buy any hardware.
I was thinking in this way to, but I was unsure if I can still use
Asterisk
On 9/25/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Adrian Marsh wrote:
I'm interested in how people are clustering Asterisk, if that's possible,
or how you might be achieving a redundant solution.
I've a single Asterisk server driving the company. Its well backed-up, and
I've a
- Original Message
From: Atis Lezdins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, September 25, 2007 2:11:10 PM
Subject: Re: [asterisk-users] Asterisk Redundancy
On 9/25/07, Philipp Kempgen [EMAIL
On 16:40, Tue 25 Sep 07, Dean Collins wrote:
A friend of mine just sent me this email - he is looking for an IAX
inbound service in Holland - any thoughts?
Voip info only has Nadiz which seems to be SIP only.
We use the following IAX providers with dutch telephone
numbers in this order of
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a
business graded installation
I'm afraid I can't give you as positive feedback as you've had from other
posters. I did quite a few installations with GXP2000s about 18 months ago, and
they've caused us nothing but
Run it with a stock OpenSER installation that will accept registrations
and acknowledge them with a 200 OK, but not actually do anything with
them.
I'm trying to avoid a PC at all in this scenario. If at all possible, I want an
ATA at one end and a SIP phone at the other, no other hardware
Hmm, this seems to describe my problem - Thanks, Bart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Tuesday, September 25, 2007 6:41 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DTMF dropping digits
In
Anyone aware of how to configure one call per line on a Cisco
7941/7961? The default behaviour is to have two calls per line button,
and this is confusing for some of my users so I'd like to be able to
have the 2nd call ring the second line button, rather than being
shared with the first. I'm
You can do this with any of the Linksys SPA series ATA's or phones, just set
Make Call Without Reg and Ans Call Without Reg to no.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Gary, if you register multiple lines with the same SIP credentials the phone
will do rollover and take care of it. (2nd call comes in on L2, etc.)
- dbc.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen
Sent: September-25-07 6:37 PM
To:
[snip]
http://lists.digium.com/pipermail/asterisk-users/2007-August/195339.html
could provide you with some answers.
Hi,
This seems nice way of sharing settings, however it wouldn't take over
calls in progress. For us, currently the greatest problem is that
whenever Asterisk crashes,
A little off topic, but SipX has built in redudancy. if it is so
important to you, you should have a look.
On 9/25/07, Atis Lezdins [EMAIL PROTECTED] wrote:
[snip]
http://lists.digium.com/pipermail/asterisk-users/2007-August/195339.html
could provide you with some answers.
Hi,
Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk
softphones. I followed the steps of how to of voip-org but always have
this error:
Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL
RealTime: Failed to query database. Check debug for more info.
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in
a business graded installation (with really traffic on not 3
calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4
I have an installation right now in a real estate/mortgage company office
with 36
David,
Yes, I'm aware of that, but unfortunately it does two calls on each
line appearance (button), so the first two calls go on line 1, and the
third will appear on line 2. I'd like to limit it to 1 call per line.
Any ideas?
Gary
On 9/25/07, David Cook [EMAIL PROTECTED] wrote:
Gary, if you
Dear List,
Has anyone found or written a status panel application, windows or
linux, that uses SIP notifies and subscriptions, to gather the status of
SIP extensions from Asterisk?
And displsy nicely on a GUI?
--
Terence C. Giufre-Sweetser
Technical Support Network Engineering
SkyMesh Pty
I'm trying to use a GXW-4008 for the first time to provide simple POTS. Is
anyone using it?
How about samples of SIP.CONF and EXTENSIONS.CONF?
Do you have advice for configuring the GXW-400x for this application?
How long a local loop will it support on the FXS ports?
When I started
One more thing i noticed today,
with SIP 2.2 and Polycom 601 i wasnt able to enable buddy watch to use with
hints.
I'll spend more time on it later to see what is up with that.
On 9/25/07, Mike [EMAIL PROTECTED] wrote:
I am having a similar issue with 4.0.0. Mine is that it doesn't get any
There have been a number of instances where recent changes in the *
code have led to a degradation of TTS in the 1.4 releases. I have no
idea whether this is relevant to ABE in general or the version you're
running. However, for a number of us the fix was to edit app_swift.c
(version 2.0rc1 from
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