I wonder if I setup a softphone on each terminal if they will actually
work as independent phones well enough, but haven´t tested it.
MF escribió:
Hi all
I'm looking for a softphone that works well under terminal services
environment,
we need to set up 24 to 32 phones for a call center,
a
ng at the L() optin in Dial. I
define a max call time, say few hours, then warn every x seconds, then cut
the call.
--
thanks,
Yusuf
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Manri
Hi,
I need to send a message to an agent when the ACD starts to ring on he/she.
I have and application already built that sends such a message (just
like a cti), just don't know how to get from asterisk which agent was
selected prior to ringing him (or during ringing), so that I can get
in
when people should be calling, but they keep calling the whole day
instead)
C F escribió:
Set the PRI cause:
http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+PRI_CAUSE
On 8/15/06, Manrique Feoli <[EMAIL PROTECTED]> wrote:
Hi, I´m in a bit of a hurry here, I need
Basically, if I do hangup on the first line the console shows:
Starting simple switch on Zap/2-1
Accepting overlap call from '' to '3423' on channel 0/1, span 1
executing Hangup ("Zap/2-1", "") in new stack.
I believe this is actually picking the call
Hi, I´m in a bit of a hurry here, I need to reject calls before
picking them up.
If I do hangup on the first line, does anyone knows if the line counts
as picked up for the Telco?
how about if I register the incoming callerid, and then do hangup on
the second line?
thanks
___
ts or
phones ringing at once or what did you mean? I need 6 phones now,
and not expecting more than 2 ringing at once. (hopefully)
cheers
Manrique
Rich Adamson escribió:
Manrique Feoli wrote:
Hi all,
I need to setup 6 phones about 3/4 of a mile from the main box,
(can't do it wit
Hi all,
I need to setup 6 phones about 3/4 of a mile from the main box, (can't
do it with VoIP yet because of networking issues), does anyone knows if
the boards can resist such a length for FXS ports.
Right now there is a Dialogic MSI160 working fine.
The actual length in a straight lin
Thanks for your reply Tony(I'm starting a new thread now, sorry about
that!!)
I see we'll have to do some developing for what I'm looking for,
but thinking again about it, I should be able to make one room for chatting of
4 users, and one room only for listening of about 120 calls, s
Hi
I need to setup a meetme room where you could accept say 120 incoming
calls to listen to the chat, BUT, only the first 4 can talk, so when
one of the first 4 leaves the room, number 5 becomes 4 and is able to
talk on the room.
Is this doable with meetme?? (am I making any sense?)
I
Hi,
I've read all over that the manager conection (via sockets) isn't good
for high traffic applications with multiple manager connections at the
same time with one asterisk, the connection hangs and many other problems.
having said that, my question is:
Has anyone worked on a fairly hig
le ),
maybe it doesn't make any sense does it?
Manrique Feoli escribió:
Maybe the question is, how can I call someone right after I something
happens, in this particular case if the Dial is not answered.
Manrique Feoli escribió:
Hi all,
I am receiving a call on one E1 and try to se
Maybe the question is, how can I call someone right after I something
happens, in this particular case if the Dial is not answered.
Manrique Feoli escribió:
Hi all,
I am receiving a call on one E1 and try to set up a call on another
E1, if the second call succeds, fine but if
Hi all,
I am receiving a call on one E1 and try to set up a call on another E1,
if the second call succeds, fine but if the second call doesn't
answer (or if the second E1 link happens to be down)I can't manage
to execute another line of my dialplan to try to setup the call via
a
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Manrique Feoli
Gerente Investigación y Desarr
why not calling them back?
1- you get the call on the 800 line, and after the operator evaluaes
the situation, dials a digit or something, the system calls back the
same number, but via your preferred route/system/billing.
2- get the call, then play an automated message to the user explaini
On 7/27/06, *Manrique Feoli* < [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> wrote:
If you have line of sight between the points, maybe you could setup a
wireless link point to point, I know some people who have done it over
3 to 5 miles range, they get 10 Mbps, (but don´t know i
If you have line of sight between the points, maybe you could setup a
wireless link point to point, I know some people who have done it
over 3 to 5 miles range, they get 10 Mbps, (but don´t know if you
could get more).
just a thought
Joe Pukepail escribió:
Fiber? Otherwise maybe look a
extensions end.
Andres escribió:
Manrique Feoli wrote:
Hey I need a quick advise here, I must be missing something basic.
I get a call from an Zap E1, and dial into a Voip extension, if the
extension hangs up first, the next line of the dialplan gets executed,
if the pstn hangs up first,
Hey I need a quick advise here, I must be missing something basic.
I get a call from an Zap E1, and dial into a Voip extension,
if the extension hangs up first, the next line of the dialplan gets
executed,
if the pstn hangs up first, shows "exited non-zero on ZAP/6-1" and
the next l
Hi, all
I have an * which receives calls from PSTN and some of them fo to an E1
where another system is working (Dialogic Boards).
I need to be able to send a signal to * from the system with the
Dialogic boards, preferrably via the E1 so that * knows it has to move
the call from slot ZAP2
an 11 supports
5ESS, but are willing to find out.
thanks
Manrique
Matthew Fredrickson escribió:
On Jul 25, 2006, at 12:53 PM, Manrique Feoli wrote:
Hi all,
Here is the situation:
A call comes in to an Alcatel PBX and it sends it to an E1 on * ,
this * either sends the call to a VoIP extensi
Hi all,
Here is the situation:
A call comes in to an Alcatel PBX and it sends it to an E1 on * , this
* either sends the call to a VoIP extension or needs to forward it to an
extension back on the Alcatel, but WITHOUT using another slot of the
E1 (no tromboning or hairpinning).
I've read
Ive done it with a tunnel set with OpenVPN, and works quite good,
there is a slight increase of lattency but not noticeable to humans.
that is doing it via UDP tunnel, we also tried via a TCP tunnel and
results weren't good, lattency increased more than desired and voice
quality was poor.
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