I would have happily bought 20 channels at $10/channel, but at most will
be buying only a single channel now :\
Pascal Bruno wrote:
Not sure if anybody noticed, but it seems like Skype For Asterisk is out.
$66 per channels, pretty pricey
My skype number appears to belong to a pool given to Level 3
Communications, and it was out of the same 1000 block as my Google Voice
number. You could block all Level 3 numbers for your area, but it would
run the risk of blocking legitimate customers from calling you.
Casey Boone
Jared
Dave Platt wrote:
OpenVPN doesn't suffer from this problem. Although it's SSL-based
(and one might think it does everything through SSL-over-TCP),
it actually sends the VPN traffic via UDP... it uses TCP only
for the negotiation and administrative aspects of setting up
the VPN connection.
just for a test, run service iptables stop as root on the asterisk
server and then reboot your phones. after that, try again and see if
the phones are making communications with asterisk.
you can turn the firewall back on with service iptables start
jonas kellens wrote:
Hi there,
this is
GrandCentral/Google Voice does just this, although I have no idea what
they use for a back end to make it happen. When someone calls your
GC/GV number, it forwards out to a list of numbers you have given the
service. You can choose to answer the call, send it on to voicemail, or
a couple of
?
What I am hoping to be able to do with this is allow for 10-15
simultaneous inbound from Skype calls, no interest at first for
receiving nor making PSTN calls via Skype.
Casey Boone
Tim Panton wrote:
On 23 Feb 2009, at 15:13, Dean Collins wrote:
Asterisk/Skype update available here
?
What I am hoping to be able to do with this is allow for 10-15
simultaneous inbound from Skype calls, no interest at first for
receiving nor making PSTN calls via Skype.
Casey Boone
Tim Panton wrote:
On 23 Feb 2009, at 15:13, Dean Collins wrote:
Asterisk/Skype update available here
I am curious as to if there are any updates on this?
Olivier wrote:
Hi,
Has anyone any return to share about Skype-Digium beta program ?
I would be very curious to know how things are going on this.
Regards
guys can we take the flame fest off list please? kthx
Douglas Garstang wrote:
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 17, 2006 1:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk
as long as they are in the same network segment as the asterisk server
you can use arp
man arp
mojowrkn wrote:
All, Can anyone point me to the best way to find the mac address of a
phone on my system?? I can get the ip's just fine but dont seem to be
able to pull mac addresses from any
you could always do a subversion checkout to a temp path and then do
search/replace courtesy a perl or a sed script (ie, replace something
like BINADDR with the address to bind to on that box).
after that rsync/cp/mv/whatever into /etc/asterisk
just a thought
Casey Boone
ShawneeLink
have anything to offer one way or the other for clipcomm and
sipura? anyone tried them both out? im leaning towards the clipcomm at
the moment just because of price and the included fxo.
Casey Boone
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you could try setting the * box to pull timing from each pri connected
to it and set the nortel to be a master for that circuit and see if that
helps any
Casey
Anthony Rodgers wrote:
Greetings, everyone, and Happy New Year!
I have a question relating to running two PRIs into a single
look into linux advanced routing and traffic control
lartc
Casey Boone
Jose Limeres wrote:
Hi all,
Has anyone a good piece of advice on using traffic shaping embeded with
*? As in our case it is not possible to configure it in the ADSL router
we would like to implement some kind of bandwidth
servers off of ebay if cost is an issue, such
as lucent max 4048s and cisco 5400s. voip just is not a good way to
carry a modem call
Casey Boone
Denis Vella wrote:
Hi,
I'm trying to use modems with Asterisk+VoIP Gateways in an attempt
at providing an Internet service.
Home_PC--Modem
had bad voice quality i have to say at least. that part has
seemingly worked rather well
Casey Boone
Ade Agbero wrote:
I have had numerous problems with Grandstream HT-386 new and old
firmware, my convidence in Grandstream is at a very low point right now.
I wish you luck,
Ade.
*/Bob
nope, i havent :\
Keith Yoder wrote:
Casey Boone escreveu:
can someone who has a grandstream handytone 488 working with making
outgoing calls through the fxo port please post the parts of their
config that deal with this port? i cant quite seem to get it to make
outgoing calls despite having
i greatly appreciate the information and will be giving it a whirl later
today :)
Casey
Soner Tari wrote:
I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount in sip.conf. On the FXO section of HT488
web admin page you enter these
.
i have been told that [EMAIL PROTECTED] has this built in to just a button
hit, but i dont want to reinstall the box and would prefer to use
asterisk directly
Casey Boone
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it and see personally, find a box you are willing to use for
an ipcop test and go to it.
Casey Boone
Mojo Jojo wrote:
We are looking for a good firewall replacement which will basically do
pot blocking and QOS.
Our current solution just plain stinks..
We basically need to handle the traffic
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