Hi everyone
Any one used Radius based accounting for asterisk.Please give me details.
Thanks
Nikhil
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Hi Friends,
I have occurred same problem on Asterisk-1.8.X. version.
I need to upgrade our production asterisk-1.6.2.6 server to asterisk-1.8.X
version.
I have already used root user during configuration and installation of
asterisk-1.8.X version but getting same error as *increase the maximum
Hello
I've never used Asterisk for a three-person call, and would like to
check that MeetMe is the way to do this.
The ADSL modem provided by my ISP offers free calls to
landlines/cellphones when using a handset connected to an RJ11 port on
the modem.
A three-person call can be set up
hi,
is there some way to balance accross sip trunks by the number of calls?
example
3 trunks
alfa(max 30 channels, priority 1),delta(60,priority 2),omega(90,priority
3)
alfa have 25 calls now
i want next call terminate to delta. how to find in asterisk the current
calls number on sip trunk
Yes, my agents dial “willy-nilly”...
I can´t use the ex-girlfriend because, the line numbers that uses the agents
are diferent. May be agent 1 today use line number 553455 and tomorrow
553461...
On Wed, Feb 2, 2011 at 3:51 PM, Danny Nicholas da...@debsinc.com wrote:
On Wed, Feb 02, 2011 at 03:29:37PM +0530, Nikhil wrote:
Thanks of reply. The command core show verbose is working. but the
problem is, for one call we can see 2 results,there is no common field
on these two.
Take a closer look at the output. The link between the 2 can be found
by
Then DISA (I had it as DASI in OP because Im working from not so good
memory) is probably your best bet. It is a simple built-in feature that
lets you get an access code in the dialplan before performing an action
such as dialing.
Check this link
http://nerdvittles.com/index.php?p=73
On 02/03/2011 11:41 AM, marek cervenka wrote:
hi,
is there some way to balance accross sip trunks by the number of calls?
example
3 trunks
alfa(max 30 channels, priority 1),delta(60,priority 2),omega(90,priority 3)
alfa have 25 calls now
i want next call terminate to delta. how to find in
In article 1296748085.2237.16.camel@shaft,
Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Is there an option on MeetMe that means the conference room is only
available if an admin user is logged in?
I've had a look the the application from the asterisk cli but I can't
really see what I'm
Mike wrote:
I was hoping to use this Queue not for professional agents in a call center,
but for reception. When the receptionist (lowest penalty) is not at the
desk, then some junior sales person can pick up those calls.
We have our receptionist setup in a front-desk queue that has 2
On Thu, 2011-02-03 at 16:39 +, Tony Mountifield wrote:
In article 1296748085.2237.16.camel@shaft,
Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Is there an option on MeetMe that means the conference room is only
available if an admin user is logged in?
I've had a look the the
We have our receptionist setup in a front-desk queue that has 2 phones in
it.
The incoming call rings directly to the phone for 30 seconds, if not
answered, plays the, Please wait while we find someone and then drops
them into a queue. At this point, it rings the operator phone again and
I think you are looking for a way to have such a report, on console:
CallCenter*CLI sip show channelstats
Peer Call ID Duration Recv: Pack Lost ( %)
Jitter Send: Pack Lost ( %) Jitter
2.168.11.731ea659 00:03:39 00236 ( 0.00%) 0.000 004685
Hi, I have asterisk 1.6.2.6 on a Debian Lenny system.
When I try to send a fax in T.38 mode I receive this error
ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel
'SIP/eutelia-sirio-out-' is in an unsupported T.38 negotiation
state, cannot continue.
In my sip.config general section I
2011/2/3 Marcello Colucci (SIRIO Informatica s.a.s.)
mcolu...@sirioinformatica.it:
Hi, I have asterisk 1.6.2.6 on a Debian Lenny system.
When I try to send a fax in T.38 mode I receive this error
ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel
'SIP/eutelia-sirio-out-' is in an
On Wednesday 02 February 2011 14:21:50 Jason Parker wrote:
On 02/02/2011 02:14 PM, Frank Liu wrote:
Hi there,
Per the instruction from http://www.asterisk.org/downloads/yum , I
setup the yum repository on my Centos 5 x86_64 machine and did a
yum install asterisk18 asterisk18-configs
Hello,
I have an installation in Austria; ISDN service provided by Austria Telekom.
The main number of the service is 6 digits. Incoming calls may contain 2
additional digits, which I then use to route the call to the correct
extension. Telekom sends me all the digits.
My problem is that when
After someone sent me an email saying his directed pickup did not work. I
realized I forgot to mention that directed pickup needs to be enabled in
extensions.conf i.e. add the following
exten=_*8.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten =
Hi Kevin,
Did you have any luck tracking down the missing rtpkeepalive code? I'm really
looking to get this working asap so I'd be happy to copy in/compile/trial some
code if there's any available.
Regards,
Ryan.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi,
I am using Asterisk 1.6.2.11 to test 3rd party Voice Message server (VMS),
currently when VMS send NOTIFY message (standalone NOTIFY, no previous
SUBSCRIBE
for the dialog for SIP), asterisk responded with 489 Bad Event, in the debug
log
it indicates as the following:
[Feb 4 13:27:06]
My outgoing FXO calls are answered but have no audio in either direction
if I have callprogress=no in chan_dahdi.conf. If I change to
callprogress=yes then the audio returns. My chan_dahdi.conf file is
listed below. Can anyone point-out why callprogress=no isn't working?
#cat /tmp/a
On Feb 3, 2011, at 11:12 AM, Kevin P. Fleming wrote:
The Queue() application can automatically pause members who fail to answer;
this would be the solution to your problem. With that solution in place,
though, the agent will still need to be able to un-pause when they return to
their
Hi All,
This posting regarding PRI voice optimization, on dahdi 2.1.0.4.
we have more than 4 machine running on 4 port PRI card with echo
cancellation hardware based.
i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now
more than 70% of call get good voice
but some of calls
Hey Guys,
I'm after a way to monitor our sip trunks (peers) and send an email if they go
down. I know I could use 'asterisk -rx sip show peers' in a shell script but
that seems messy, especially since I'd like to monitor it fairly closely (so
I'd like to run it every 20 or 30 seconds or so).
On 02/04/2011 01:34 AM, Ryan Tucker wrote:
Did you have any luck tracking down the missing rtpkeepalive code? I'm really
looking to get this working asap so I'd be happy to copy in/compile/trial some
code if there's any available.
Yes, it was lost during a merge of code into Asterisk trunk
Dear,
Meetme is a default conference application, but you can try conference or
konference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Konference
Thank you Kevin.
Today I will upgrade my asterisk and then I will inform you about the
result.
Bye
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Kevin P.
Fleming
Inviato: giovedì 3 febbraio 2011 22.10
A:
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