?
Probably because the files are not in your TFTP root. This is probably
because you are not using these files to autoconfigure your phones.
Stephen R. Besch
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman
here should be, but there isn't. The "blinking" phone
designers should have put those stupid blinking red leds - that only
flash on boot up - under the message button and flashed the display
during boot up. But they didn't and we'
on was intended to be the
telephony apparatus, not just the card. If you make that assumption,
then those who are successfully using more than one card per CPU are
getting a really good deal. The second, third ... card is essentially a
freebie.
Sincerely,
Stephen R. Besch
ng a firmware
update, you may still be able to get the phone operational if the
bootloader code is intact (which is not that unlikely), but you will
need to get a TFTP server located at the address that the phone is
attempting to access.
Hope this helps.
Stephen R. Besch
is a good possibility that the
firmware is not the problem. It's just something that you must consider
when a releas version of the firmware works and the beta version does not.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://l
pening?
thanks in advance
Well, 1.0.5.18 is Beta firmware.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
ly.
2)Copy the appropriate subject line into your message subject before you
send the message so that we can actually tell what you are writing about
without having to search through your reply.
Stephen R. Besch
___
Asterisk-Users mailing list
[
no reason that you can't assign a public and
a private IP to the same NIC. In fact, I'm doing exactly that on both my
Windows and my Linux boxes.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com
n a
private network - they are not routable by design. I had quite an
argument with Grandstream about this when I first purchased the phones.
As a result, the firmware was modified to accept a null router entry for
use with private IP ranges.
Sincere
visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
This all reminds me so much of Jonathan Swifts bit about the BigEndians
and the LittleEndians (referring to which is the 'correct' end to open a
soft boiled egg) in Gulliver's travels.
Stephen R. Besch
___
How do you downgrade the Budgetone to 10Mb? I don't see anything on the
configuration page to do that. Also the specs on the Budgetone say it
is a 10Base-T port.
You can't. It already is 10MB. It can't do 100MB.
Stephen R. Besch
___
tone generator for free - see their web site.
Also there is at least one script available from the * mailing list.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
ports using a small 4-port switch
(maybe 40 bucks or less) you don't even need much extra wiring. The
packet traffic level from an IP phone is just not enough to be of any
concern unless you are moving gigs of data simultaneously over the same
shared port.
Stephen R.
y new TSU-600 for $99.00.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
defined in the server that lie
on your private network
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailma
untime (available from MS for free) and
that you have SAMBA running on the * server.
Sincerely,
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or upd
tory do not have the time/date
embedded in the audio (".gsm" or ".wav"). The time/date is in the ".txt"
file that accompanies the message. When * plays the message, it gets the
date/time from the ".txt" fi
figure if you can find out
what IP address the phone is using for TFTP and put a TFTP server on
that IP. Even if the IP is public, you could still use it temporarily as
long as your machine and the phone are isolated from the public network
on a private switch/hub.
Ste
Michael George wrote:
On Wed, Oct 20, 2004 at 01:46:01PM -0400, Stephen R. Besch wrote:
I have never been able to get the Grandstream to register reliably -
with any version of the firmware.
So you mean you don't use the Grandstreams, then?
On the contrary, I use almost nothing but GS p
registration on the phone. It's useless anyway with fixed IP and just
reduces reliability (as you have discovered the hard way). Asterisk
periodically sends polling packets to the phone, so it will know when it
is reachable and when it is not. And, the phone will still authenticate
against t
at I
would know)...
My understanding is that if you don't ask for or need support, RedHat
(and the other distro's as well) can be downloaded and installed for free.
Am I wrong about this? Isn't that the whole point of the GPL on linux?
Stephen R. Besch
___
le level. As far as I can tell, the phone delivers its audio to
the sound card over the wav interface, so be sure that the volume and
mute settings for wav playback are also set correctly.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
other words, to
get your impedance matcher to work, you will need to match impedance
over frequency in such a way as to eliminate any unexpected phase shift,
otherwise cancellation will not be improved. In fact, if you are not
very careful, you may just m
or the existing
T1 lines. Just run a T1 crossover cable from the bank to a spare port in
one of the Digium cards.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or upd
f you don't believe it, hook a scope up to one of the active
interrupt request lines on the bus. It's very revealing.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
quot;, that the
student had failed to make the distinction between a fallen woman and a
woman who had merely fallen.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
Ryan Courtnage wrote:
Had not seen anything on "Read" anywhere else, must have been looking
in all
the wrong places. this is a simple solution to my problem.
FYI - Read() is described on the wiki:
http://voip-info.org/wiki-Asterisk+cmd+Read
___
Even b
Michael George wrote:
I just got a Budgetone 101 and I have it hooked to my * box. I thought I'd
read somewhere that we can program the buttons on these phones to send DTMF
tones, thereby effectively programming them.
However, according to the user's manual, they have predefined SIP
functionality.
this subject. Note that it is possible to retrieve
voice mail on an unregistered phone, but you won't be able to receive
calls (because * won't know the IP address).
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://l
=SomeCleverPasswordorOther
etc.
etc.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
And here I was trying to figure out how to kill the blinking display :-)
OK - dumb newbie award hereby rewarded to me. Thanks. And I had already
checked the wiki and done what you suggested in sip.conf - so my
stupidity wasn't total :-)
Stupidity may be a bit strong in any case. The real stupi
f you use
internet for a voice channel, make sure that you use a separate NIC to
talk to the RAID, and assign it a lower interrupt priority than those
used by the telephony hardware.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
pts are listed
in a table in the user manual (p32).
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Andrew Kohlsmith wrote:
On Tuesday 20 July 2004 18:18, George Pajari wrote:
In spite of what my learned colleague implies above, there is more to
Canada than Ontario (Bell's territory).
Please retract your statement that I implied anything of the sort; I never
even mentioned the province I was in
I now have a batch mode installer for GSConfigure which works on at
least 2 of the machines here. If you are having trouble getting the VB
installer version to work, give this a try. It's at
http://www.acsu.buffalo.edu/~sbesch
Stephen R.
sources of GSConfigure.
that could save a lot of
time. I have no intention of competing with gsconfigure since I think
it's an excellent
What? Competition is good!
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Maron Kristófersson wrote:
and the attachment is here :)
Maron Kristófersson wrote:
I'm very close to making this work in the crossover wine emulator on
linux. Currently I am getting an error when trying to download the
config directly from an ip address. See attached snapshot for details.
Mar
Steve wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 08 July 2004 05:45 pm, Stephen R. Besch wrote:
The most recent version of GSConfigure is available at
www.buffalo.edu/~sbesch Several serious bugs that kept the program from
getting started have been ferreted out and corrected
Maron Kristófersson wrote:
OOPS! Ignore that last post, I found the snapshot in your next post.
SRB
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://
ristófersson
Stephen R. Besch wrote:
Greg Boehnlein wrote:
On Thu, 8 Jul 2004, Neil Cherry wrote:
Stephen R. Besch wrote:
The most recent version of GSConfigure is available at
www.buffalo.edu/~sbesch Several serious bugs that kept the program
from getting started have been ferreted out and corr
Remco Barende wrote:
On Thu, 15 Jul 2004, Chris Glover wrote:
On Thu, 15 Jul 2004 [EMAIL PROTECTED] wrote:
Note "The previous method for terminating IAX2 calls using Connect! will
cease to be available at midnight (GMT) on August 15th, 2004." The
message I got was at 1:51 AM EST. That means I wa
Greg Boehnlein wrote:
On Thu, 8 Jul 2004, Neil Cherry wrote:
Stephen R. Besch wrote:
The most recent version of GSConfigure is available at
www.buffalo.edu/~sbesch Several serious bugs that kept the program from
getting started have been ferreted out and corrected with the help of
Bruce
The most recent version of GSConfigure is available at
www.buffalo.edu/~sbesch Several serious bugs that kept the program from
getting started have been ferreted out and corrected with the help of
Bruce Komito. The program is now actually running on someone's machine
other than mine. I have bu
Stephen J. Wilcox wrote:
I was wondering about that too..
Following the instructions on that page for config did not work for me. Setting
up a config file like the sample one made no difference to the phone (I can
confirm it did tftp it okay). Also the method references md5 checks and I dont
see
g a lot of typing if the
phones don't all have the same password. The only time a password is
requested is when it can't be extracted from the configuration file for
some reason.
The new version (1.0.25) should be available shortly at
http://asterisk.4gurus.or
Tomas Prybil wrote:
Stephen R. Besch wrote:
snip
This is the install package for the program. Running setup will
install the program into Program Files\SachsLab\GSConfigure and put a
shortcut in the start menu under "Phones". The sources are installed
to the application directory i
to everyone. Even if you have guest
disabled, this still leaves you vulnerable to snoops discovering your
configuration. With that in hand, they can make phone calls on your
dime. I would change the access rights on all of these files to 640.
Stephen R. Besch
___
Stephen R. Besch wrote:
I've just finished a general purpose configuration utility for the GS
phones:
1) Generates files from scratch (using MAC), from HTML config
listing, or by directly downloading from the phone.
2) Does multiple simulteneous edits.
3) Can reboot as many
uld brief me on the GPL issue, and (perhaps someone
else) offer a distribution point, it's free for the asking, VB sources
and all.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-
Kyle Hagan wrote:
Brian Buhrow wrote:
Hello. I've seen this behavior. What happens is that the
Grandstreams forget to continue registering with Asterisk after a
while. I
bet when you find this happening, that sip show peers doesn't show
ext/ext
ip address for the one that isn't working.
address. So, for example, if 192.168.10.100 requests ring1.bin, TFTP
will get ring1_192.168.10.100 and serve it up as ring1.bin, effectively
lying to the phone. Then you only have to generate your ring files and
place them in the TFTP root folder with access rights
out the problem, has verified on my * server, has
indicated that they will fix it. It just requires patience on our part.
The fix is apparently not a very high priority - true really, since it's
failure is merely a convenience issue of not having to wait for the last
key timeou
evertheless, there really doesn't seem to be any good reason to go to
the 1.0.5.0 version anyway. It hasn't fixed any of the outstanding
issues (at least those related to use with "*", or added any really
useful functionality.
Stephen R. Besch
ter option is a really serious, and
idiotic, mistake. They really are obliged to fix this as soon as
absolutely possible.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUB
Stephen Rosebush wrote:
I just got myself a GS HandyTone and it works great, it was a breeze to
setup. My only issue is I seem to be hearing a humming noise on the line
when I am in calls..
You have a short (or leak) to ground somewhere in the analog line.
S Besch
___
Tony Hoyle wrote:
Stephen R. Besch wrote:
Not as far as I know, at least not exactly the way you have outlined
it. Try this:
1. call comes to you
2. You hold the call and call other person.
3. You say "Someone wants to talk to you, OK, thanks"
3a. Other person the
at you are transferring the call
3d. You transfer the call using the transfer feature on the phone
4. You hangup and first person is transferred to other
person?
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.
.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
it rejects it later
for being overly long.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
nge of 002935 and up. It would be
interesting to know if the MAC's from the phones that work are
significantly higher than these.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asteris
erver. I have tried it
on at least three different phones, purchased in 2 different lots and
still no luck. Maybe the phones just don't like me.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailma
else has had to do the same.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
ht
resolve the issue. If anyone can fill in the remaining fields, it would
be cool.
Hope this helps.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update option
Thomas Gallaway wrote:
Brian Capouch wrote:
Thomas Gallaway wrote:
My ringtones just work on all the grandstream's :-)
Do the "URLS" for the ringtones at the top show up as something other
than all zeroes?
I've fiddled with this until blue in the face, and the ring sounds
just like the ring it
Thomas Galloway wrote:
Stephen R. Besch wrote:
Duane wrote:
Grandstream v1.0.4.68 firmware
Am I missing something obvious about the new ringtone feature? The
4.68 firmware updates as usual from my TFTP server, the new version
shows up in the phone's web page, but the ring tones, while prese
Thomas Gallaway wrote:
Jeremy McNamara wrote:
Stephen R. Besch wrote:
Duane wrote:
Grandstream v1.0.4.68 firmware
Am I missing something obvious about the new ringtone feature? The
4.68 firmware updates as usual from my TFTP server, the new version
shows up in the phone's web page, but the
Jeremy McNamara wrote:
Stephen R. Besch wrote:
Duane wrote:
Grandstream v1.0.4.68 firmware
Am I missing something obvious about the new ringtone feature? The
4.68 firmware updates as usual from my TFTP server, the new version
shows up in the phone's web page, but the ring tones, while prese
page, all show a version of 0.0.0.0, and all
functionality regarding them is disabled. Are we maybe jumping the gun
here a little bit or is there something special about getting them to load?
Stephen R. Besch
P.S. Grandstream, if you are listening, then Early Dial is still broken!
It's been m
ll potential of 2.2V per cell (13.2V for a standard 12V
car battery at 25 degrees C). Typically, the optimum charge voltage
would be about 13.8V for a 12V battery.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.
sts.digium.com/mailman/listinfo/asterisk-users
Make sure that you don't have either the "Use # as dial key" or "Send
Flash Event" options set to yes. They must both be set to "NO" for
transferring to work on the GS
Stephen R. Besch
___
You guys are funny! Mark Spenser! Haha! I knew immediately it was from Mark
Musone!
Wasn't Mark Spenser a medieval poet or something?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRI
Iain Stevenson wrote:
I'm running 4.50 because of adverse reports of 4.53 etc.
Is abbreviated dialling (aka Early Dial) working yet - it's been out of
commission for most firmware from 35 - 50 releases.
No. Tested it last week on '4.54 - still broken - but somewhere along
the line they fixed th
server IP in here! Use default port
registration_expiration=10
You may find registration to be a problem with the GS. See comments above.
send_dtmf=in-audio
This must match the entry in sip.conf (In the GS world, in-audio =
inband)
Sincerely,
Stephen R. Besch
__
wing:
SIP Server: 192.168.0.102
Outbound Proxy:
5) I would set this to be the same as the server if you want to make
outbound calls.
Hope this helps
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailm
host=dynamic
disallow=all
allow=alaw
allow=ulaw
context=intern
secret=
mailbox=113
dtmfmode=rfc2833
nat=0
My budgetone is set to send DTMF via via RTP (RFC2833) with a payload
type of 101. (tried 100 and 102)
The first 2 codecs are set to PCMU and PCMA (tried to switch thos
n you place the call is that,
obviously, your end has no trouble determining when you have hung up,
regardless of setting that affect the detection of the remote end
hanging up.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
ement when extensions
change, but the simplest solution is to instruct users to include the
extension number when they record their name for the directory.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailma
files sorted into chronological order
when they are renamed?
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman
nt" option set to "YES"
in the GS configuration. If you do, flash will not be sent as a SIP
event and the flash button won't work.
Stephen R. Besch, Ph.D.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mail
mber?
I've been banging my head, my eyes are blurry, and Google isn't answering...
Could someone just point me in the right direction.
Thanks
-Art
If you're using the GS101, save your Excedrin, it doesn't display alpha.
The number is all you wi
handset.
On my phones, there is already a notch on the handset. Only the
appropriate "bump" on the cradle is missing. A strategically placed
2-56 socket head screw solves the problem nicely.
Stephen R. Besch
___
Asterisk-Users mailing l
some
applications. However, it comes close enough for many purposes.
For reference, I'm using a slightly modified version of the stdexten
macro in the dialplan (no t or T) and either SIP INFO or INBAND on the
phones.
Stephen R. Besch
___
Aste
Olle E. Johansson wrote:
Stephen R. Besch wrote:
Olle E. Johansson wrote:
Going back to the subject, what does the grandstream really do,
SIP-wise, when you press
the transfer button?
4.3.7 Call Transfer The user can transfer an active call to a third
phone by using the “Transfer” button. The
with “Also:”
200 ->
From Transferee to Recipient
INVITE ->
<- 100/180/200
ACK ->
<- RTP Media ->
I have no idea if this is accurate, I just copied it and replaced the
arrows indicating direction with
cting my channel bank to the
Digium T1 card. I repaired the cable and the mysterious disconnects went
away. It may be way off base for your case, but it might be worth some
investigating.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROT
connected one of the phones tip to ground or ring to ground.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
Chris Albertson wrote:
--- Steve Underwood <[EMAIL PROTECTED]> wrote:
A power spectrum plot will tell him he has a 60Hz hum. I think he
already knows that. I think he can definitely consider solutions
without
following your suggestion. :-)
No, It's not a "60Hz hum". Yes, 60Hz is getting int
what
ever happened?
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
when mysterious
things are happening to me that are not happening to everyone else. The
bad news is that this kind of thing can be almost impossible to find. I
consider myself really lucky.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL
s are NAT'ed
By the way, contrary to many posts on this list, I have
disallow=all
allow=ulaw
allow=alaw
appearing only once in the general section and all (22) of my phones and
HT-286's work perfectly - just another one of those great * mysteries I
guess.
Stephen R. Besch
Andrew Kohlsmith wrote:
If you're getting echo of your own voice, but the remote is getting a
clear signal, then Asterisk echo cancellation is working properly. It
is the remote provider not echo cancelling properly.
I don't buy it. If that were the case then why would I not _also_ get my
own
all
allow=alaw
allow=ulaw
Tim
Tim,
Look harder in the mailing list and at the WIKI. There are literaly
hundreds, if not thousands, of posts on this exact issue.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.
, problem now fixed.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Steven Critchfield wrote:
On Fri, 2004-02-06 at 13:46, Stephen R. Besch wrote:
Every once and a while * throws a new wrinkle at me. It has started, all
on its own, to make these annoying little beeps evey time a message
prints at the CLI. If I bring down * and restart, they go away for a
time
* is starting to experience the "Terrible Twos"! No one else seems to
be complaining about this, but I nevertheless assume that I can somehow
disable this feature, I just can't seem to find out how. Maybe
something like
>CLI stop beeping d
uickly, is
easy to apply and is electrically and chemically inert.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.c
Stephen R. Besch wrote:
I just fetched today's cvs (1/30/04 11:10:31). Compiles/installs on my
test machine (ASUS A7V, 900 MHZ). However, If I try to compile on my
production machine (Elite K7S5A, 2.4GHz, 512MB) while * is running the
zaptel and asterisk compiles seg fault. I am assuming
Jon Pounder wrote:
nail polish and liquid-paper work fine for this sort of stuff.
Too brittle. Adhesion to metal and many plastics is marginal. Fine for
places where there is no shock (of the physical kind). If this is
earthquake territory, stick to the silicone or ty-wraps.
Stephen R. Besch
a flakey circuit board by removing the sockets and soldering in all the
(formerly) socketed chips. The square pin spring contacts in those
connectors are only designed for a few insertion/removal cycles. If
that is the case, you should get a good repair tech. to replace the
this source is not well corrected by
the * echocanceller.
Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman
1 - 100 of 177 matches
Mail list logo
