Here's a cheap solution for PoE piggybacked over your existing network.
http://www.amazon.com/gp/product/B0002R6X9S
On Jul 14, 2013, at 3:12 PM, bilal ghayyad wrote:
Hello;
We have a cisco switches but they are not PoE and we need only to have PoE
device so the cables come for it first to
On Oct 30, 2010, at 2:28 PM, Zeeshan Zakaria wrote:
My main asterisk server is under unusual heavy attack, and so far
Fail2Ban has blocked about 30 IPs, from various different countries.
At this time it is blocking about 1 IP address every few minutes.
Just wondering if anybody else is
On Jul 21, 2010, at 7:05 PM, Apu Islam wrote:
Can any good men on this group share me the firmware of a Cisco 7960 Phone?
Currently this one has Call Manager Firmware installed, I am trying to
convert it into SIP.
Much appreciated.
Apu
Try google keywords: index of P0S3-06-3-00.bin
On Jun 4, 2010, at 8:40 AM, Danny Dias wrote:
Hello Asterisk users,
I'm having a little problem with an Asterisk installation on Ubuntu, i had
installed many asterisks on CentOS but never in Ubuntu, the problem is that
Asterisk and DAHDI does not start at system start...i have to make
On Jul 6, 2009, at 10:00 AM, Jerry Geis wrote:
Over the weekend I tried to migrate a system from
asterisk 1.4.25, libpri 1.4.1 ZAPTEL 1.4.12.1
to
asterisk 1.4.25, libpri 1.4.7 (not 1.4.1) and DAHDI 2.2.0
I removed all old zaptel by:
mv /etc/zaptel.conf /tmp
mv
On Apr 23, 2009, at 3:14 PM, Rick Dwyer wrote:
Hello list.
I posted this over on the Biz section but some of the members thought
I might find more people running Asterisk on the Mac over here.
Here's my question:
I have looked at PHLink and PhoneValet and neither seem to be able to
do
On May 21, 2008, at 9:55 AM, Sanjay Rajdev wrote:
We are using Asterisk 1.4.13 on FC6 and have a T1 card with 20 DID's.
We are wanting to use one of the DID's for Fax, is this possible or
do we have to add some addition Hardware and what is the best way to
do this.
I know that similar
Jerry,
I'd imagine that you can achieve this through SIP Event Notify, via
AGI using
sipsak (www.sipsak.org)
I'm doing a similar thing with Cisco phones, and it works great.
Here's an example of what I pass to the phones.
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
From: sip:asterisk;tag=2427962554
On Mar 11, 2008, at 3:25 PM, Jerry Geis wrote:
I have a situation when a T1/PRI line comes into box 1
then uses SIP over to box 2 and all my phones are on box 2.
if the person is not at their desk on ring no answer I am calling
their
cell phone
which places the call back over SIP to box
On Mar 9, 2008, at 1:34 AM, Daniel Suleyman wrote:
Dear all, interesting behaivior of the Read function.
I have SIP phone(XLITE) attached to my Asterisk.
SIP.conf
[7007]
type=friend
qualify=900
host=192.168.85.27
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=alaw
allow=ulaw
Goran Dj. wrote:
Maybe trivial question, but I cannot find an answer:
How to autostart Asterisk (daemon) on Slackware 10? I know that I should
put something in /etc/rc.d, but what?
In my /etc/rc.d/rc.local
# Put any local setup commands in here:
/sbin/ztcfg
/etc/rc.d/rc.hdlc
/usr/sbin/asterisk
the latest CVS.
Any ideas would be greatly appreciated.
Niles Ingalls
I'm using a Wildcard T100P, and have 11 incoming lines.
zapte.conf
span=1,0,0,esf,b8zs
loadzone=us
defaultzone=us
fxsls=1-11
zapata.conf
; Zapata telephony interface
; Configuration file
[channels]
musiconhold=default
language
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