Have TE406P with echo cancellation $1,999 new.
With warranty.
Mark
Voice International
www.voiceinternational.com
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[EMAIL PROTECTED] wrote:
Have TE406P with echo cancellation $1,999 new.
With warranty.
This is a non-commercial mailing list (it's right there in the mailing
list address line). Do not post ads here for any purpose.
It might also serve you slightly better to use a subject that has some
Dear helpers and supporters!
I have been playing with the A2billing for a week, but I still get stuck on
creating a working trunk to terminate the call. Is any one show me how to
setup a trunk in A2billing or pointing me to some where that I can find out
the information.
I would very
Carlos Chavez wrote:
On Fri, 2005-12-16 at 12:48 -0800, Dave wrote:
I had a lot of issues with 480i too and this is how I
resolved it:
1) Make sure that the file on the tftp server is
called firmware followed by the type they suggest (I
do not remember the type name)
2) Once this is done,
Can we only advertise used stuff? I haven't seen anyone else that's selling
stuff get told not to post ads. VoipSupply comes to mind.
j/w
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Friday, December 16, 2005 3:57 PM
To:
On 12/16/05, MapsAir [EMAIL PROTECTED] wrote:
Dear helpers and supporters!
I have been playing with the A2billing for a week, but I still get stuck on
creating a working trunk to terminate the call. Is any one show me how to
setup a trunk in A2billing or pointing me to some where that I can
Hi all i have some problems with my pbx and asterisk codecs.
if i use g711u or g711a codecs. the line never hangup. and the origin
and destination are connected until i restart my pbx or asterisk
But if i use GSM all work fine.
is possible to solve this problem? or use only gsm codec?
--
.-
Hi all i have some problems with my pbx and asterisk codecs.
if i use g711u or g711a codecs. the line never hangup. and the origin
and destination are connected until i restart my pbx or asterisk
But if i use GSM all work fine.
is possible to solve this problem? or use only gsm codec?
In article [EMAIL PROTECTED],
Ross C [EMAIL PROTECTED] wrote:
Can we only advertise used stuff? I haven't seen anyone else that's selling
stuff get told not to post ads. VoipSupply comes to mind.
Ads are perfectly welcome on the asterisk-biz list: that's what it is for.
But not on the
Ross:
This is the -users list, where ads are not allowed. Ads are allowed, and
are welcomed on the -biz list. VoIPSupply.com adhere's to the communities
rules; we post for sale items on the -biz list.
Garrett Smith
[EMAIL PROTECTED]
716-250-3408 Direct
716-903-9495 Cell
Can we only advertise
On Fri, 2005-12-16 at 18:28 -0500, [EMAIL PROTECTED] wrote:
Ross:
This is the -users list, where ads are not allowed. Ads are allowed, and
are welcomed on the -biz list. VoIPSupply.com adhere's to the communities
rules; we post for sale items on the -biz list.
and infact they just did to
Ok. Thx :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter aka
Bret McDanel
Sent: Friday, December 16, 2005 5:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DIGIUM
On Fri, 2005-12-16 at 18:28
Hi:
Is there any possible way to make TDM01B answers when
the other side pick up the phone ,and to prevent it
from answering just when it starts ringing?
Regards;
chawki
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Hi,
i have two [EMAIL PROTECTED] 2.2 server. i want if one of my asterisk server down. other is taken control of my first server and call goes through.
Is it possible in asterisk.
Usman
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On Fri, Dec 16, 2005 at 05:08:07PM -0600, Rich Adamson wrote:
Hi all i have some problems with my pbx and asterisk codecs.
if i use g711u or g711a codecs. the line never hangup. and the origin
and destination are connected until i restart my pbx or asterisk
But if i use GSM all
I think you have to run the update_database function. That code was
written over a year ago and has not been touched since that I'm aware
of. I suspect the Friends support should be moved over to realtime but.
Darren Wiebe
[EMAIL PROTECTED]
Insider KT wrote:
Hi. I am having some
Hi,
i have two [EMAIL PROTECTED] 2.2 server. i want if one of my asterisk server down. other is taken control of my first server and call goes through.
Is it possible in asterisk.
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Hi,
i have two [EMAIL PROTECTED] 2.2 server. i want if one of my asterisk
server
down. other is taken control of my first server and call goes through.
Is it possible in asterisk.
Is this Malik Muhammad or Steve?
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Hi,
Asterisk High Availability Solutions
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
SteveSent: Saturday, December 17, 2005 10:14 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users]
Hi
check this URL will help you for this
since this list only asterisk.. not [EMAIL PROTECTED]
http://sourceforge.net/forum/forum.php?thread_id=1398290forum_id=420324
ram
On 12/17/05, snacktime [EMAIL PROTECTED] wrote:
On 12/16/05, MapsAir [EMAIL PROTECTED] wrote: Dear helpers and supporters!
Steve Kann wrote:
Richard Scobie wrote:
My SVN asterisk systems use the following topologies:
1) PolycomSIP - *1 -IAX- *2 - H323 Gateway
2) PolycomSIP - *1 -IAX- *3 - Zap TDM400 Analog
3) H323 Gateway - *2 -IAX- *3 - Zap TDM400 Analog
There's a few points in here so far:
1) the new
So how many are you trying to display? A hundred is pretty easily done.
On Dec 16, 2005, at 3:03 PM, Terry H. Gilsenan wrote:
Kerry Garrison wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Terry H.
Gilsenan
Sent: Friday, December 16, 2005
hi all,
What is the best Asterisk-compliant for Dell machine is recommended? I
will have roughly 400 users in a production office. thanks!!
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50 extensions, 27 trunks, 1 queue, any tips would be great appreciated,
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones
Sent: Friday, December 16, 2005 8:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
hi Steve Totaro,
They are monitoring the SIP packets also, so if we changed the ports
they are starting to block the traffics.
--
Thank You,
Code Lover
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Excuse me Chris!
Forgive me that I don't understand what you are really mean?
I would very appreciated if you let me know some think about the rules, and
how we would get help from people and how to find some previous information
that has been posted from someelse before that we may need it.
Lan wrote:
Forgive me that I don't understand what you are really mean?
He means that this is the list for Asterisk Support - [EMAIL PROTECTED] is a
completely different product. (Asterisk is simply the PBX software,
where as [EMAIL PROTECTED] is a complete distro that just happens to include
Hi
i managed to install asterisk and h323, i am facing few problems, please
help me
1.) i have setup the LD_LIBRARY_PATH like the following, but i have set it
again when i reboot the server, how to slove this issue.
PWLIBDIR=$HOME/pwlib
export PWLIBDIR
OPENH323DIR=$HOME/openh323
export
On Sat, 17 Dec 2005 09:06:50 +0300, Code Lover wrote:
They are monitoring the SIP packets also, so if we changed the
ports they are starting to block the traffics.
ipsec is the only way to get around this. Or iax2?
/Niklas
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The best Dell for a production environment Asterisk server is no Dell
at all. They make some great workstations, but I've had many problems
with their servers(as have many others on this list) when trying to
use them in production for Asterisk. Take a look at the Digium
compatibility list:
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