Displaying 20 results from an estimated 4000 matches similar to: "Operator Panels?"
2004 Dec 15
2
chan_sccp compile problem w/ CVS head?
Any ideas? I edited the Makefile as instructed, ty.
Now compiling .... sccp_channel.c 279 lines
sccp_channel.c: In function `sccp_channel_send_callinfo':
sccp_channel.c:48: structure has no member named `callerid'
sccp_channel.c:49: structure has no member named `callerid'
sccp_channel.c:49: structure has no member named `callerid'
sccp_channel.c:49: structure has no
2006 Mar 29
1
OT: FOP and reverse_transfer
When I drag and drop a call from the PSTN to a SIP phone (the SIP phone's
icon) using FOP .25 with * 1.0.9 with the intent of transferring it, the
called party gets transferred rather than the calling party. This is
controlled by the reverse_transfer parameter in op_server.cfg but the
behavior is exactly the same whether the parameter is set to 0 or 1. This is
after the call is picked up by
2006 Mar 29
7
Reporting?
Is there anyway in asterisk to figure out how much time an agent has
spent on the phone? I know I can see total time for a call (inbound
or outbound) but where/how do I view queue stats?
2005 Jun 23
4
Monitoring Sirrix quad BRI channels
Hi all,
How are things going ?
Is there a way for me to individually identify each BRI channel on the Sirrix quad BRI board.
The reason I ask is because our client uses the "Asterisk Flash Operator Panel" to monitor its external lines and transfer calls from the lines to the various SIP phones.
The "Flash Operator Panel" requires that we set a static value for each line or
2006 May 09
3
Announcement: FOP 0.26 released
I'm pleased to announce that Flash Operator Panel 0.26 has been released!
FOP is a GPL'd switchboard type application for the Asterisk PBX. It
runs on a web
browser with the flash plugin. It is able to display information about
your Asterisk box in real time. It is included in FreePBX,
Asterisk@Home, DeStar, startShop, and several other projects both free
and commercial. You can grab the
2004 Aug 20
6
Sipura endpoints
Anyone have experience with Sipura's? Anyone know if they offer a
warranty? Would like opinions on these, good or flame.
We bought *one* to test with and it died, can't even get a
response from Sipura "support". Could anyone recommend another device to
replace these? Prefer 1 or 2 port design.
Ty :-)
2004 Aug 16
2
Problem compiling chan_sccp
Hi,
I recently bought a 7910. I found out too late that it would not do
SIP as I initially thought. Anyway before ditchingit for a 7960 I
wanted to try it out, I read that the guys at
http://chan-sccp.sourceforge.net/ had done some improvements to the
original chan_sccp driver and having 80% functionality with this
model.
I have not been able to compile their driver and keep getting the
2006 Jun 24
5
ASTCC: How to reset periodically all "card in use" flag back?
If a user calls and hangs up before the destination party rings, than
the in-use flag remains set! This is one case, but maybe there are many
other cases.
I have created a number the user can dial to reset this flag. However,
that is written in the manual!!! Who reads a manual anyway!!!!
I want to make to reset all in use flag with a program. Has anybody done
it, or has a better idea?
My idea
2005 Feb 01
4
astGUIclient users should not upgrade to Asterisk 1.0.5
Hello,
Just confirmed this on my end, because of the massive changes that have been
made to callerID handling in asterisk 1.0.5 many of the features of the
astGUIclient suite will not work on this new version. The latest stable
version recommended is Asterisk 1.0.3. We will work on trying to find ways
around the new callerID rules that the asterisk developers have put in place
and hope to have
2004 Sep 16
3
Creating conference calls from within Astman.
Dear All,
I have a requirement to 'originate' a number of calls to various external
users from within a conference room, so that the end users does not pay for
the call.
I know that within Astman I can define an extension and then originate the
call from that extension. Can I define a conference room (how would I
configure that on astman? What channel would it use?) and then generate a
2004 Aug 24
7
SMP Performance
We're looking at implementing Asterisk in our department in the near
future, we're looking at anywhere from 15-25 extensions. The machine we
were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/
1GB of ram. I've heard bad things about running Asterisk on SMP
machines? Would we be running into any performance issues with this
machine?
Tim Jackson
Network Engineer
2005 Mar 15
6
Realtime config
Having problems getting realtime working, I'm trying to use odbc for all
of this. I've got Fedora 3 and have been fighting with odbc for a day
now. I think I got it working correctly, however I can't seem to get the
realtime portion working. In asterisk 'odbc show' shows it connected, I
see it on my (odbc) mysql server connected and all, it connects and just
idles. So, without
2005 Feb 12
3
Is there a Caller ID issue in the latest CVSStable
Nicol?s Gudi?o <asternic@gmail.com> wrote:
>>> Paul, 1.0.5 stable suffers from caller id issues as well, at least for
>>> SIP channels. What fixed things for me was swapping in app_dial.c from
>>> 1.0.2 stable (didn't try others). You could also just diff app_dial.c
>>> between versions to find the problem but I took the lazy way out the
>>>
2005 Mar 24
14
Realtime mysql problem?
All, I get this whenever trying to dial to a peer when the peer
registered to another server. I'm basically trying to use realtime to
check for the peer and dial it.
Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f",
"IAX2/brak-test/107") in new stack
Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM
sip_users WHERE name =
2008 Jan 17
1
AddQueueMember and Flash Operator Panel
Hello users!
Recently I read that AgentCallbackLogin is going to be deprecated soon.
Wanting to set up a few callback type queues, I set them up as suggested
in queues-with-callback-members.txt.
I was able to set the queues up completely this way, however, I'm trying
to use Flash Operator Panel (aka AsterNIC) to monitor the agents' login
status. FOP monitors their status if I call
2004 Sep 21
3
chan_sccp/SEP<mac>.cnf.xml
HI all:
I have spent a large amount of time configuring/installing phones
connected to Asterisk. Halfway through the process I discovered that my
Cisco7960 with 2 7914 expansions was not supported in the SIP protocol.
After reverting to SCCP 6.0(4.0) I am now perplexed with the hassle of
configuring SCCP to properly work with Asterisk.
So far I have gotten the phone to dial and receive calls
2004 Sep 15
3
Cisco 79xx + asterisk + some functions Q
Hi,
I am new to Asterisk and have some general questions _before_
I start buying equipent to install and get everything up-and-running.
(this means I have no running Asterisk (yet)).
I have read already a lot of doc, but some things are not
clear to me, since I'am inexperienced in Asterisk and PBX.
The goal:
Make PBX using Asterisk and Cisco 79xx equipment for 25
phones (1 x 7970, 4 x 7960,
2005 Feb 14
1
Flash Operator Panel - lots of problems
On Tue, 15 Feb 2005 03:02:45 +0100, Stefan Gofferje
<stefan@gofferje.homelinux.org> wrote:
> Hi folks,
>
> I have some trouble with the FOP and would appreciate if anyone could
> point me into the right direction.
There is a FOP user list, although not too active.
http://www.asternic.org/
> Is there a way to define a button like Zap/g1/6000 and have it light up
> when
2006 Apr 21
1
Flash Panel / Queue Slots
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Hello,
is there any way to make the Flash Operator Panel show which agents are
logged in in a specific queue? (both static and dynamic agents)
I've played around with the queue / queue agents settings from the Flash
Panel documentation (http://www.asternic.org). The way it is described
there, I could only make the Flash panel show that a queue
2005 Jun 07
1
CallerID/chan_sccp
When sending a call to a line defined on chan_sccp,
there is an error on the console that says:
Jun 7 08:22:29 WARNING[3924]: sccp_channel.c:79
sccp_channel_send_callinfo: Incoming call SCCP/Line1-00000008 doesn't
have CallerId name
Is this because of the changes in the callerid name from stable to head?
Or does someone know of a way to correct this?
--
respectfully, Joseph