o. Oh
yeah, and we were on a T1 PRI, which is not *supposed* to have echo.
Unfortunately when I left the company, they finally replaced the phone
system to get rid of the echo and customer complaints.
A motherboard list would be REALLY great, indeed.
Best regards,
Ryan T
AM). How in the heck would/should I go about figuring out
what the interrupt service latency or the PCI bus latency is doing. Any
other thoughts on the front? I'm using GS phones so maybe their echo
can algorithms are to blame... hmmm...
Here's to hoping,
Ryan Thrash
_
On Jul 23, 2004, at 1:22 AM, Andres Junge wrote:
What is a RMA?
Return Merchandise/Materials(something like that) Authorization.
It's a number from the mfr, that when the product arrives with it on
the box, tells them to expect some dead hardware.
rt
_
er on the
molex connector on the card?
-- Ryan Thrash
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ling out as has been discussed here previously ad nauseum
with no one being able to really figure out its source. I wish I knew
where to really start poking around to try to help get to the bottom of
this.
Best regards,
Ryan Thrash
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dual Xeon 2.4s and a Supermicro
SuperWorkstation 7033A-T (X5DAL-TG2 motherboard
http://supermicro.com/products/motherboard/Xeon/E7505/X5DAL-TG2.cfm ).
Echo training=800 on a recent CVS helped, but did not totally resolve
the issue.
Best regards,
Ryan Thrash
__
implemented the
echotraining=800, but it's still there. We haven't touched TX/RX gain.
We can also give anyone access and a SIP account if that would be
helpful.
Best regards,
Ryan Thrash
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htt
peakerphone button prior to
placing the receiver on the hookswitch. When you pick up the receiver,
just press the hold button again to resume your call. I too found out
the hard way.
HTH,
Ryan Thrash
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I'm not aware of any way of doing this currently, but this has made it
to the planning board of Voicemail3... the timing for which is
unfortunately undetermined at the moment.
HTH,
Ryan Thrash
On May 25, 2004, at 11:14 AM, Bruce Komito wrote:
When a user has voicemail, I would like * to
Your English is just fine. :)
What's your extensions.conf and sip.conf for your Grandstreams look
like?
What are your options in the GS config webpage for:
1) NAT transversal (and are you behind a NAT firewall)
2) Send Flash event
3) Send DTMF
Best regards,
Ryan Thrash
On May 3, 2004,
I can verify that snom 200s will support up to 5 line appearances and
you can happily change back-and-forth between them.
Now actually successfully transferring those calls when more than one
call is in those line appearances is another thing entirely, when using
the soft keys or the transfer b
I would also offer feedback that we too have random calls with echo on
our end, that can't be traced to a reproducible event. It's very odd
and can be frustrating, as it's a big distraction for those that don't
know better. Like a bad cell phone connection when you hear yourself
talk. For us, t
FYI, with 1.0.4.55 and NAT set to off (but with the * config set as
nat=yes), I'm able to bypass stun servers completely with a GS phone as
well.
HTH,
Ryan
On Apr 18, 2004, at 5:08 AM, Brancaleoni Matteo wrote:
you don't need stun to make GS work under NAT
with *
Just set NAT=yes into the GS, a
I *think* the default password is "" (all zeros).
HTH,
Ryan
On Apr 17, 2004, at 10:38 AM, WipeOut wrote:
Pertti Pikkarainen wrote:
There is a way.
Right after reboot, and when you see the first text, hit any key
and you will get a 'boot menu'. Give the phone an ip-address and
define a tft
First, check zapata.conf to see what is in there. Next, I've not heard
of any luck with the name portion on T1s, but the number can be changed
for us.
HTH,
Ryan
On Apr 16, 2004, at 10:30 AM, Mike Machado wrote:
You can usually get CLI on an E&M robbed bit T1 by configuring it
right.
Instead
On a Grandstream ATA and CVS HEAD from last night, and with echo off,
I'm able to receive faxes. With echo on, no go.
HTH,
Ryan
On Apr 15, 2004, at 1:58 PM, Ariel Batista wrote:
Osvaldo Mundim wrote:
Hi,
Does anybody have ATA 188 working with any kind of fax machine? I've
tried many different c
Would this fix also help random quality issues both on a LAN and also
with a remote SIP based installation running CVS from 3/22.
We're having too frequent complaints in conjunction with Grandstream
phones and very stuttery/choppy sound, usually outgoing to land lines,
to the point of being uni
Could you post a link?
Thanks!
On Apr 13, 2004, at 2:59 PM, Nick Knight wrote:
Just a quick note, I have been putting together a TAPI driver for
Asterisk, this enables the user to perform things like click to dial
from any TAPI enabled app (such as outlook or ACT etc). At the moment
it
is very
The GS phones do not currently support conferencing on the phones using
the conference button. You'll probably have better luck setting up a
conference room, help with which I'm absolutely worthless... The
on-phone conferencing should be addressed in a future GS firmware
revision.
HTH,
Ryan
On
On Apr 9, 2004, at 9:52 AM, Tilghman Lesher wrote:
On Thursday 08 April 2004 22:41, Ryan Thrash wrote:
Scenario: a person selects an Auto Attendant option that fires off
the Directory application (CVS circa 3/22). Three questions:
1) How do they escape if they didn't mean to go there in the
Let's say an unsuspecting soul accidently selects the Directory option
from an Auto Attendant (CVS circa 3/22). Three questions:
1) How do they escape if they didn't mean to go there in the first
place (without having to hang up...)?
exten => 1,1,Directory(vertex)
exten => 1,2,G
Sent 12 hours ago and it never showed up (slightly reworded here).
Sorry if this is a duplicate:
-
Scenario: a person selects an Auto Attendant option that fires off the
Directory application (CVS circa 3/22). Three questions:
1) How do they escape i
You should be able to do a reload, not having to restart (and bringing
the system down).
On Apr 8, 2004, at 8:48 AM, Jain, Sonal wrote:
Is it true that every time we make a change in the configuration file
we need to restart the asterisk server. This will not be practical in
the production env
Remove the semi-colon in front of "[global]"
HTH,
Ryan Thrash
On Apr 7, 2004, at 8:15 PM, Jeremy Bogan wrote:
Sounds like an error in your config file. Want to paste the contents
in? Thanks...
Sorry:
;[global]
;hostname=localhost
dbname=asterisk
password=
user=asterisk
;port=3306
Wow... talk about a detailed response; thanks!
In our situation, we've got a T-1 voice PRI from Allegiance Telcom. For
the benefit of those of us who aren't as in the know as you are (and
who have no affiliation with a CLEC), is there a way to be able to
control what gets sent out as our name p
uot;, "1") in new stack
-- Executing VoiceMailMain("SIP/122-7161", "2142618000") in new
stack
-- Playing 'vm-login' (language 'en')
Ideas?
Thanks,
Ryan Thrash
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I think Xten is included in Lindows, but I could be totally wrong.
Probably tweaked to work out of the box with that Distro...
Xten is really pretty good.
The one I know of is X-Pro/X-Lite from http://www.xten.com/
I doubt that there is a Linux version available...
Markus
I contacted X-Ten a
I'm running into a similar situation. We have 3-digit extensions and a
4-digit DID numbers that get used for for outbound CID. Therefore, no
$CALLERIDNUM direct access to voicemail. Suggestions?
What do you do when $CALLERIDNUM of the caller isn’t the 4-digit
extension? I set all of my users
We had an issues with an Intel Zero Channel hardware RAID controller
that wouldn't allow us to install either Fedora Core 1 or 2, so we
couldn't test with *. Given that we didn't try to convert our 9 to
Fedora, either. We got it running great under RH 9.
HTH,
Ryan Thrash
On Ap
going to be somewhere without
cell service, you could change the number over the voicemail line.
Thanks,
Ryan Thrash
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Actually, ignore that... forgot to take the check the calendar pill
this AM. Doh!
rt
On Mar 30, 2004, at 11:46 AM, Ryan Thrash wrote:
How did the launch meeting go?
rt
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How did the launch meeting go?
rt
On Mar 29, 2004, at 1:36 PM, Steven M. Sokol wrote:
The VON show has started off with a number of interesting
announcements.
First among these is a big announcement from Pingtel that they have
created
a not-for-profit corporation called SIPFoundry. This new c
Does register_globals need to be on to work with this? And if so, any
chance that will be turned off in the (hopefully near) future?
Thanks, Ryan
On Mar 24, 2004, at 9:09 AM, Areski wrote:
I just finished an other version, all my apologies, cause I made it for
mysql then I ve done the change to
I can't seem to find an answer in the archives covering this (or maybe
I just missed it)... Setting up * and hope to accomplish the following:
1) Use 5 of our DID numbers from our PRI for inbound fax reception
2) When * receives a call on one of these lines, it digitizes the
incoming fax to a mu
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