Displaying 20 results from an estimated 3000 matches similar to: "SetGroup and CheckGroup"
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
2005 Jan 24
1
SetGroup and CheckGroup problems
I have a rather long dial plan, but it includes support for call waiting.
However, the setgroup checkgroup commands don't seem to be working. Can
anyone help on this one?
Excerpts are below. First exten-vm is dialed and then dial-new.
As I understand, priority 1 increments the active channels for the caller
and then in "dial-new" priority 8 increments for Arg3, or the Callee
2004 Jun 25
1
Howto: Use setgroup, checkgroup to check incoming and outgoing client limits
Hi there,
I was wondering how I can use setgroup and checkgroup for perfom incoming
and outgoing limitation checks.
I've have some users that doesn't what to be able to recieve more than 1
call at a time, and I also want to limit a users outgoing call abilities.
Any help would be greatly appreciated.
Kind regards
Cf
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus
2003 Nov 25
3
Handytone 286 - calling out
Hi,
Just received recently released Grandstream handytone 286 ATA for
testing.
I can call ATA from any other extensions and conversations seems to be
of quite good quality. However placing calls from ATA is not possible at
all to any extensions.
After dialing there no indications of any kind from ATA at all. It just
"hangs in there".
ATA is behind NAT, registers to an * with public IP
2003 Dec 17
12
128 kbs satelite link
Hi all,
Anyone has experience using * through
128 kbs (or bigger) satelite link?
In particular I am interested to hear how many calls could be put
through 128Kbs satelite link simultaneously?
Ta
SJ
2006 May 04
0
SetGroup and CheckGroup. Need some help on the dialplan
>From this list I found that I could use SetGroup and CheckGroup to do
what I wanted. But I'm not quite sure how I do it.
The case is that I have 3 user groups, and one main group. The main
group is for all the incoming calls from external phones. The main group
should be allowed to have 3 calls at the time.
The 3 user groups are internal groups that I want to limit by ony having
one
2004 Jan 09
12
USA dial plan
Hi,
Do the callers in USA dialling from USA Telco lines always have to
prefix the CITY/AREA code with "1" in order
To successfully make a call to other USA destinations?
----
I have not been to USA (yet) :)
Ta
SJ
2004 Oct 01
1
Agent Login Problems
See comments below.
Henry Devito wrote:
> Here's the problem. When I call 555 to login, it asks for the agent
ID
> which I enter as 501, it asks for the password which I enter as 1234,
> then it asks for the extension I dial 501 It then says that extension
is
> not valid. What am I missing? Of course 501 is valid I can make and
> take calls from it now.
>
>
>
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
2004 Aug 11
1
limit incoming calls to sip extens
Hi all,
I've been using the following method to limit calls to sip clients to 1:
exten => 200,1,SetGroup(200)
exten => 200,2,CheckGroup(1)
exten => 200,3,Dial(SIP/200)
exten => 200,103,Busy
This works fine for a single extension.
However, I also need to dial groups of sip clients. It appears that SetGroup can only be used once per channel.
This (useless) example would not
2004 Aug 31
2
limit the length of extensions
How do I limit the length of an extension? In my test IVR/Automated
Attendant (whatever it's called), at the beginning it plays "if you know
your parties 3 digit extension, you may enter it now) and then it gives
a list of options. If the caller puts the 3 digit extension, it goes
through fine, if they press 1, or 2 it goes to the selected menu option,
but if they dial 91235551212 it
2004 Jan 08
5
AbsoluteTimeout Users Messages
Hi, All
Is there a provision for "AbsoluteTimeout" application to notify
called and calling party of the reason why the call suddenly ended?
This way, the parties will be much better informed, hence they
will/should not think that
their VOIP/telco provider(s) are providing bad service.
Ta
SJ
2005 Jan 06
1
Re: Asterisk-Users Digest, Vol 6, Issue 73
Hi John, Kevin, Tor and Wiley (and everyone else) -
>> I guess the phone just doesn't register as busy when there is only one
>> call on a line. It has to have two calls on a line appearance to
>> register as busy. Has anyone figured out how to disable this hold
>> feature and just have the second call go to the second line, the third
>> call to the third line,
2005 May 10
1
Limiting outbound calls
My VoIP provider allows me to have more than one call outbound on the
same line simultaneously, for some reason. I am pretty sure that they
do not want this to happen, so I'd like instead to limit each line to
one call.
I do not want the users to have to dial another prefix to go out on
another line. Is there any way to add multiple accounts for my _9.
extension and have Asterisk
2005 Jun 19
2
outgoing call routing
I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip
extensions and a regular phone connected to the box. All routing works fine
from the regular phone connected to the box, whether its going to FWD,
broadvoice or the PSTN. The problem I am experiencing comes from making
calls from the sip phones. They get routed correctly to the sip and iax
trunks but when making calls
2006 Mar 31
4
How to check if a phone / line is used?
In the past I used SetGroup and CheckGroup to figure out if my allowed
providers lines are all used or not.
Since most of my provider have given me a single line anyway, I wonder
if there is a way to check if this (provider) line is taken already.
How can I do that?
Same is with the phone. How can I see in CLI if a phone is now in use or
not?
"Sip show peers" shows me just if it is
2004 Sep 20
2
H.323 call problemm (no sound)
Hi all,
I'm having trouble with H.323 outbound calls, * connects but there is no
sound in both ways.
I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which
using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729
licenses installed and this is onli one call.
I tested my * with another ITSP over SIP and G.729 codec and there was
all ok
Here is my configs
2003 Dec 31
6
Happy New Year!!
Hi all,
Let me be the first to wish everyone, especially the Digium team, an
awesome year in 2004..
Later..
2005 Aug 31
2
detecting extensions in use
Hi all,
We've got a department that has 5 phones using a * 1.0.9 box. They need
to have an extension that rings all 5 phones at the same time. Getting
all of the phones to ring isn't a problem, but they are running into a
problem with the phones ringing in their ears when they are already on a
call.
Example:
Caller one calls the queue, all of the phones rings, and employee one
2004 Jun 10
2
BT is moving to IP ONLY
Hi, all
This is certainly very good news!
http://www.neowin.net/comments.php?id=21119&category=main