-index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk
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Aaron Johnson
Star Networks
Main: 602-889-3000 | Office: 602-889-3002 | Cell: 602-741-4660
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By going to http://www.grandstream.com/y-286.htm and clicking on
Product Data Sheet I was able to determine that the device does
support silence suppression. You will want to ensure that this feature
is turned off as it can cause some of the problems you have been talking
about.
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Aaron
with Speex is finding good
phones that support it. Our clients mainly use Polycom and Cisco phoes,
which do not support Speex.
--
Aaron Johnson
Star Networks
Main: 602-889-3000 | Office: 602-889-3002 | Cell: 602-741-4660
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.
--
Aaron Johnson
Star Networks
Main: 602-889-3000 | Office: 602-889-3002 | Cell: 602-741-4660
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Asterisk wrote:
Is there any way of knowing how many zap channels were in use at a
particular time ?
We are getting a lot of circuit busy and reorder from our Sip
phones when we're making outbound calls, and I'm convinced that we
have not used our full complement of 30 channels (EuroIsdn
I didn't look into any disconnect fees yet. That's a good one to be aware
of since the phone appears to be available in a bundle with their service.
I had a look at the Pulver cordless. How does it (at $199) compare to the
Zyxel 2000W (~$250 from voipsupply.com)?
Adi
They are the same phone.
Rich Adamson wrote:
Ok. Thanks a lot anyway. BTW, do you know how many g729 licenses I
need in this situation? Maybe 1 is not enough. Maybe I need 2: 1 for
decoding and one for encoding.
RODOLFO
As the documentation states, you need 1 license for each instance of
g729 in use. That
Nabeel Jafferali wrote:
I recently bought a bunch of IP500s and before shipping / tax
they were $170 / each (including power supply). We are lucky
to have received such a great discount, but there's no reason
to pay more than $200 for an IP500.
How do the Polycom IP500/600 phones compare to
Puddle wrote:
We'd like to use VoIP Phones, and possibly Software
Based phone (*NIX/Windows enviroment).
Personally, I would recommend that you stay away from any software based
phone. It is my experience that whenever a computer gets under any kind
of load, it tends to degrade the voice
[EMAIL PROTECTED] wrote:
I am trying to put together a list of 'departments' to request as
voice prompts. I have the biggies (sales, accounting, shipping,
etc...) but I want to make sure I do not miss any. If anyone anyone
has some suggestions (Ha... that is like going to an NRA meeting ans
Matthew Boehm wrote:
We recently switched to 729. I wouldn't expect that to cause built-in
conferencing to stop working.
Matthew
- Original Message -
From: Kevin P. Fleming [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday,
Marcello Lupo wrote:
Hi to all,
we have a community of people on an * box that use SIP softphones to talk each
other. Can you suggest me the quickest and simple way to let someone know who
is online without have to call one by one the persons to look if they are
present or not?? Something the
Currently I am having 2 issues with my Avaya 4602 phone:
First, the phone registers with my Asterisk server, but when I start
dialing I get a busy signal after 4 digits. I specified in the dialplan
on the phone to expect 10 digits and that solved that problem, but I
still immediately get a
I ran SIP debug on the asterisk server and this is what I got. It looks
as the avaya phone is trying and retrying to register, even though the
phone shows that it is registered. I'm stumped.
___
Aaron Johnson wrote:
Currently I am having 2 issues with my Avaya 4602 phone:
First, the phone registers with my Asterisk server, but when I start
dialing I get a busy signal after 4 digits. I specified in the
dialplan on the phone to expect 10 digits and that solved that
problem, but I still
Aaron Johnson wrote:
Aaron Johnson wrote:
Currently I am having 2 issues with my Avaya 4602 phone:
First, the phone registers with my Asterisk server, but when I start
dialing I get a busy signal after 4 digits. I specified in the
dialplan on the phone to expect 10 digits and that solved
I have written two new wiki pages.
Avaya: http://www.voip-info.org/wiki-Avaya+4602+configuration?page=Avaya
How to upgrade and configure an Avaya 4602:
http://www.voip-info.org/wiki-Avaya+4602+configuration
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Tenorio, Leandro wrote:
Just guessing, but 've you tried the to rename Sip_4602ap1_0.ebin to
appsip.ebin
I did. The problem turned out to be with my HTTP server. I switched
HTTP servers and everything is now running fine.
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Aaron Johnson wrote:
I attempted to update an Avaya 4602 phone to the latest SIP firmware
and now the phone stops at the bootloader. It keeps requesting an
appsip.ebin file from my HTTP server and is no longer checking my TFTP
server for update files. Since no appsip.ebin file was included
I attempted to update an Avaya 4602 phone to the latest SIP firmware and
now the phone stops at the bootloader. It keeps requesting an
appsip.ebin file from my HTTP server and is no longer checking my TFTP
server for update files. Since no appsip.ebin file was included in the
firmware update
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