Displaying 20 results from an estimated 100 matches similar to: "groupcount fraud problem"
2010 Jun 09
0
1.6 how to use groupcount and counteronpeer in queues to avoid ringinuse
Dear all
i'm planning an upgrade of some asterisk installation from 1.4.32 to
1.6.0.28 (as i think it should be the most stable now).
Reading the UPGRADE-1.6.txt file i've noticed that:
* SIP: The "call-limit" option is marked as deprecated. It still works
in this version of
Asterisk, but will be removed in the following version. Please use
the groupcount functions
in the
2010 Sep 27
0
groupcount - show usage
hi,
i'm using groupcount to limit max calls in pbx
i want show/graph made calls usage
it's possible make this from cli/ami?
something like
asterisk>group show usage
name channels
group1 3
group2 5
google doesnt help
thanks
---------------------------------------
Marek Cervenka
=======================================
2008 Aug 21
3
After Dial execution, using DIALEDTIME, ANSWEREDTIME
Hi,
I noticed that when dial terminates it does not return to the dialplan,
and therefore can not execute any entry after Dial().
Is there any trick to overcome this limitation ?
How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if
I can not execute anything after Dial()?
I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls
end
2009 Oct 30
2
DAHDI/ZAP overlap dialing
Hi,
I have a PRI euroisdn link between an Alcatel PBX and Asterisk.
I'm having some trouble with overlap dialing.
Suppose I dial '874053' from an Alcatel extension ('7034') where '87' is an Alcatel prefix of type "ARS Prof.Trg Grp Seiz.with overlap".
I'm expecting Asterisk to receive '1004053' (where '100' is a prefix which always shows
2010 Aug 17
1
MySQL Connect problem...
Right, I'm baffled.
I have:
exten => s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2)
exten => s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\
(caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\
VALUES\ (\'${CALLERID(number)}\'\,\'${ARG1}\'\,NOW()\,NOW()\,\'Yes\'\,12))
exten => s,n,MYSQL(Query RESULT1 ${DB1} SELECT\
2007 Oct 30
1
Size of Exten when using IAX
Hi,
We are use IAX protocol between two asterisk servers.
Now we send information through this protocol by using EXTEN
We see that the variable EXTEN only holds 66 characters.
If we set a value larger then 66 characters, for example 70 characters.
The last 4 characters are cut off.
Is there a way to increase this variable?
Kind regards
-------------- next part
2005 Sep 20
0
Aterisk App ICES Question
I have a question about the Asterisk Application ICES. I've got Asterisk
setup to accept a phone call and call the ICES app which sends it to an
Icecast server.
exten => 1,1,SetGroup('stream')
exten => 1,2,GetGroupCount()
exten => 1,3,Ices('contrib/${GROUPCOUNT}-ices.xml')
exten => 1,4,Hangup
Everything works fine. Unless I have more than 24 phone calls being
2004 Jul 30
1
FW: Limit incoming calls to SIP Channels
Hi All,
Can someone please tell me how to limit incoming calls to SIP channels using
the SetGroup & Checkgroup command. I don't want any call waiting on SIP
channels and you are somehow meant to be able to do it with these commands.
Many Thanks
Daniel Niasoff
2008 Mar 28
2
wrong extension status when call-limit=1 is used
Without call-limit defined, when a sip extension calls
another sip extension then "show hints" shows that
both are InUse (as expected). When one of them hangs
up, both hints status become "Idle" (as expected).
With call-limit=1 for each SIP extension:
the caller is always Idle while the callee is InUse.
Is this behavior normal?
Doesn't sound right because if, during the
2005 Mar 14
2
Has anybody experience with SetGroup / CheckGroup commands?
I am checking on the SetGroup / CheckGroup commands, but I have some
troubles to undestand the examples.
SetGroup(moh) can be moh anything as I like? Usually moh stands for
"music on hold"
CheckGroup(1) checks if somebody in in group "moh". Does it mean I can
only have one SetGroup(xxx) ??
When I look at example 2 than I see two SetGroup commands and one
CheckGroup
2004 Jul 19
0
*** Asterisk Sun/Monday News: Time to download, Scotty!
This week starts with the exciting news: We're getting close to
Asterisk 1.0 again. After the failed attempt earlier this year,
we've been able to remove a lot of the MAJOR/CRASH bugs from the
bug tracker and Mark feel's it's time to target 1.0 again.
At this point, the community needs to work as a community,
spending extra time on finding bugs, solving issues, improving
2007 Sep 18
6
Limiting Simultaneous calls
Is there a way to limit simultaneous calls. I like to limit
simultaneous outgoing calls as more than few simulataneous calls are
charged by my voip providers. However, I do not want to have any such
restriction for internal calls.
Thanks
Jim
2006 Jun 04
6
fine-tuning asterisk questions
I have asterisk running more or less ok but I would like to turn off
call waiting and be selective about the incoming sip connections. This
is running asterisk 1.2.8 with a fxs and fxo card and a configured voip
(sip) line. Currently I'm using freePBX 2.1.1 to configure asterisk.
Problem 1) if someone is on the phone already and another call comes in
for an already engaged extension I
2013 Aug 14
2
proxy: get rid of redundant log-informations
Hi
login_log_format_elements = user=<%u> method=%m rip=%r %k
is it possible to get rid of the "proxy(test at testserver.rhsoft.net): started proxying to 127.0.0.1:143: " part
because on a proxy-only server i know that and it is explicitly not listed in "login_log_format_elements"
as well as for the "TLSv1 with cipher DHE-RSA-CAMELLIA256-SHA" it would be
2004 Sep 12
1
SetGroup Limitation!!!
Hi all,
I am just scratching my head trying to work out a way to use SetGroup to
check busy status on a sip to sip call.
The complication is that one call can't be in two groups so I have got no
way of setting busy status on both the calling and called party.
Has anyone got a way around this.
Thanks
Daniel
-------------- next part --------------
An HTML attachment was
2009 Sep 17
2
limit concurrent calls on trunk supporting multiple DID
Hello guys,
I've one SIP trunk that support multiple DID. Only the trunk is
documented in sip.conf (called DID is taken from the sip-header in
real time).
I would like to limit the number of simultaneous calls on each DID. Is
there a way to achieve this ?
My understanding is that the SIP configuration parameter
"limitonpeers" will limit at the trunk level, right ?
Thanks in advance
2006 Apr 08
6
How to set busy
For multiline phones how do you set SIP channels to busy. For instance
if SIP/101 is on a call then dial would return busy. Right now it just
starts ringing on line X, and stacks up from there.
What would be really great is if I could control how many calls by the
context. So if a call was routed via
[overload] Then the ext wouldn't report busy it would just keep ringing
available
2009 Feb 26
3
call-limit on a per destination basis
Hello,
I use asterisk to to IAX2 trunking between London POP & Reunion Island pop.
I would like to know if it's possible to do a kind of call-limit (i.e.
restrict to XX) channels but on a per dialcode and / or destination basis.
For example:
[trunk]
; reunion proper, i want to send no more than 24 channels
exten => _0262XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN})
; reunion mobile, i want
2007 Nov 30
3
How to setup redundant SIP peers
Hello list,
I try to setup an asterisk-server with different SIP-Peers to PSTN.
The Peer are working and configured in sip.conf:
[peer1]
type=peer
host=10.10.10.1
[peer2]
type=peer
host=10.10.10.2
Now dialout is no problem. Extensions.conf says:
exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30)
But how can I setup a failure-route if the SIP-Proxy "peer1" ist not
2004 Nov 20
1
Asterisk dead but pid file exists - gdb asterisk core.13089
Dear ALL,
Any clues or tips for the following gdb messages.
[root@localhost asterisk]# uname -a
Linux localhost 2.4.22-1.2115.nptlsmp #1 SMP Wed Oct
29 15:30:09 EST 2003 i686 i686 i386 GNU/Linux
localhost*CLI> show version
Asterisk CVS-HEAD-09/22/04-11:19:09 built by
root@localhost on a i686 running Linux
[root@localhost asterisk]# gdb asterisk core.13089
GNU gdb Red Hat Linux