statistics ---
34 packets transmitted, 33 received, 2% packet loss, time 33024ms
rtt min/avg/max/mdev = 64.993/121.282/229.137/42.306 ms, pipe 2
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features that
will help with the echo and clipping and if so, how much improvement
should we expect ?
Thanks.
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the web interface and checked the firmware
version on the basic tab. It was then at x.22.
I saw was in the above statements because the user interface was
different on the x.18 firmware and I am recalling from memory.
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/mailman/listinfo/asterisk-users
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and hooked into their
wireless networks with no problems.
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the * server is working as a simple NATing box, you can add * back
into the mix and then you'll know the issue you are seeing is * and not
something else.
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traversal support. I'm
using SER with their NAT helper module, it allows the phones to
connect
from behind most NAT devices. It's not a 100% solution. There're
still
cases where we need to do port forwarding on the NAT.
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Kim Lux, Diesel Research Inc
Comments below.
On Fri, 2005-01-28 at 08:18 +0800, Leo Ann Boon wrote:
Kim Lux wrote:
I was expecting to have to port forward too and yet our setup doesn't
require it, not on the laptop nor on the wireless router.
I think as long as the SIP clients open a port on the NATing device
. If you do, turn it
OFF.
Hope that helps!
- Pedro
VoIP by TRACI.net
On Thu, 27 Jan 2005 08:53:02 -0700, Kim Lux [EMAIL PROTECTED] wrote:
I'm testing a bunch of stuff before we implement our system.
I've got a SIP account and Grandstream phones. We haven't started using
asterisk
was in the above statement because the update fields are
arranged a bit differently in x.22 and I am going from memory when I
speak of the x.18 fields.
On Wed, 2005-01-26 at 09:00 +0100, Robert Rozman wrote:
- Original Message -
From: Kim Lux [EMAIL PROTECTED]
To: Asterisk Users Mailing List
mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
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I updated to firmware version x.22 and this and a few other problems
were fixed. I was running x.18 and it allowed me to do a successful
upgrade via http.
On Tue, 2005-01-25 at 08:10 -0700, Kim Lux wrote:
Are you saying that you are running firmware X.22 and it is not doing
the callback
provided by firestarter
Grandstream BT100 phone with firmware version x.22
I'll provide a HOWTO (done when I can get to it) if anyone is
interested.
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kim Lux
Sent: Tuesday, January 25, 2005 8:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk calls back after phone call
I'm connecting to a commercial SIP
http://www.timesonline.co.uk/article/0,,2-1454225,00.html
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version 1.0.5.18
BTW: If I connect the Grandstream to the wireless router directly (ie
with an Ethernet cable) it works fine. Actually, it works better than
fine: I am very happy with the voice quality and call handling.
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Kim Lux, Diesel Research Inc
BTW: When is babyTel going to get termination in Saskatchewan. 50% of
Alberta residents came from Saskatchewan and another 25% from the NS and
PEI. My Dad, father in law and uncle will all need babyTel accounts.
Kim
On Mon, 2005-01-24 at 23:33 -0700, Kim Lux wrote:
The following question
oops... wrong recipient...
On Mon, 2005-01-24 at 23:37 -0700, Kim Lux wrote:
BTW: When is babyTel going to get termination in Saskatchewan. 50% of
Alberta residents came from Saskatchewan and another 25% from the NS and
PEI. My Dad, father in law and uncle will all need babyTel accounts
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on the grandstream phone. If I pick up the
receiver
again and then hang up, the PBX starts calling me back and when I
pickup
and listen, there is silence.
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+
ethernet, and then your sip phone will just talk to the network like
normal...
Just my thoughts ...
I never knew you could set up a bridge (between two network devices)
anyway other than NAT.
I'd love to hear how this is done if you care to share.
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Kim Lux, Diesel Research Inc
Is this what you had in mind: http://bridge.sourceforge.net/faq.html
It is built right into the 2.6 kernels.
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cards don't allow spoofing of the source address. It is a
firmware restriction with some chipsets. You might find some information
in the bridge mailing list archives to help.
paste ends
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their mail server.)
I wonder if shaw or telus people lurk on this site.
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and
other governmental agencies
for uncompetitive behavior.
I think it should work.
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before the finding was released, they changed their mind. Much to the
detriment of New Zealand internet...
So, once again, the big boys get to keep their toys, while we can but
beg for quality service...
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phone ?
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but you can have
the capability now via its built in SIP capabilities.
Is this a wireless USB phone ? Does it support SIP and could it be used
to connect to any SIP server ?
Does anyone have experience with these ?
Thanks.
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Kim Lux, Diesel Research Inc
, 2005-01-12 at 19:09 -0700, Kim Lux wrote:
http://www.pcphoneline.com/skype
The VPT1000 is NOT a simple last generation USB phone audio device but
is rather a next generation integrated gateway and USB phoneset with
simultaneous dual mode Skype and SIP calling support. Skype is not
forecast
phone or a wifi
SIP phone ?
Is anyone using the Pulver WiSIP phone ? Any comments ?
How about the zyxel ?
Thanks
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An unflattering zyxel review:
http://slacker.com/~nugget/asterisk3.php
I can't help but think my questions are out of place on this list... I'm
asking questions about SIP phones and everyone else is talking about
asterisk. Sorry.
On Wed, 2005-01-12 at 20:08 -0700, Kim Lux wrote:
My wife
to a WiFi phone.
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: Kim Lux
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, January 12, 2005 5:49 PM
Subject: Re: [Asterisk-Users] Looking
min/avg/max/mdev = 21.586/24.910/40.073/4.134 ms, pipe 2
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PBXs that way.
Although I haven't completed our asterisk installation yet, the biggest
hurdle seems to be getting the VOIP to PSTN connection.
Thanks
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www.babytel.ca
Does anyone have experience, good or bad with this provider ?
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connection ? Is it necessary ?
Thanks.
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