I have this Error Please Help me
loader.c: Error loading module 'cdr_radius.so': libradiusclient-ng.so.2:
cannot open shared object file: No such file or directory
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Hi
We're getting requests coming in for higher quality audio in our call
recordings. We currently use MixMonitor and everything is being saved in
it's native 8000Hz, 16 bit wav format.
I have seen information on using Monitor and specifying a conversion to
mp3 when the call ends and the 2
On Tue, Feb 8, 2011 at 5:09 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
We're getting requests coming in for higher quality audio in our call
recordings. We currently use MixMonitor and everything is being saved in
it's native 8000Hz, 16 bit wav format.
I have seen information on using
On Tue, 2011-02-08 at 05:40 -0600, Sherwood McGowan wrote:
On Tue, Feb 8, 2011 at 5:09 AM, Ishfaq Malik i...@pack-net.co.uk
wrote:
Hi
We're getting requests coming in for higher quality audio in
our call
recordings. We currently use MixMonitor and
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Tuesday, February 08, 2011 6:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call Recording audio file quality
But if you are getting calls all the way on VoIP then you can have calls in HD
audio using HD audio codec on all locations (Server and Client). In that case
you either need use some available 3rd party solution which uses packet
capturing to trace the calls and record call using packet capture
That answer was pretty much what I was expecting. Just wanted to make
sure.
Glad to be of service :D
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On Tue, Feb 8, 2011 at 6:01 AM, fai...@vopium.com wrote:
But if you are getting calls all the way on VoIP then you can have calls in
HD audio using HD audio codec on all locations (Server and Client). In that
case you either need use some available 3rd party solution which uses packet
Hello,
you have to install radiusclient-ng
http://developer.berlios.de/projects/radiusclient-ng/
Regards
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Hi All,
I'm having some troubles with using call files.
I'm trying to establish the following:
- want to use call files to connect two (outside) extensions
- want to use the outbound routes set in FreePBX
- want to set the outgoing callerid for both calls
- want to set a custom CDR field in
Thanks much everyone for the great help. I did go through the last
suggestions about the callfile (no CRLF issue, permissions are 644 and
file owned by root, starting asterisk through strace, etc.), but none
helped.
However, by chance, I happened on a pattern: The callfile is handled
only if I...
Yes. The technology need to be used on LAN switches is port mirroring or
line tapping
-Original Message-
From: Sherwood McGowan sherwood.mcgo...@gmail.com
Sent: Tuesday, February 8, 2011 7:34am
To: Asterisk Users Mailing List - Non-Commercial Discussion
Why don't you use single callfile and set CLI and other perameters in dial-plan
as unique as you need?
-Original Message-
From: Tamás Dajka tda...@gmail.com
Sent: Tuesday, February 8, 2011 7:45am
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call files error
Hi All,
On Tue, 08 Feb 2011 14:23:12 +0100, Gilles codecompl...@free.fr
wrote:
However, by chance, I happened on a pattern: The callfile is handled
only if I...
1. Stop Asterisk through its init.d script
2. Mv the callfile
3. Start Asterisk through its init.d script
It also works if I launch Asterisk
Hello Gurus,
Can i add ${HANGUPCAUSE} in CDR after the Dial command using h extension? I
want to add the Hangup reason of call in userfield of CDR.
Regards,
Shariq Khan
0333-3501125
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yep..that would be what i said, using the nifty slang my peeps use in the
datacenters
I just wanted to be cool like them...*hangs head*...
great...now I gotta transfer to another school...
LOL, have a good one mate!
On Tue, Feb 8, 2011 at 7:23 AM, fai...@vopium.com wrote:
Yes. The
${HANGUPCAUSE} value is available on h extension.
-Original Message-
From: Shariq Khan shariqrazak...@gmail.com
Sent: Tuesday, February 8, 2011 8:30am
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] ${HANGUPCAUSE}
On 8 Feb 2011, at 13:30, Shariq Khan wrote:
Can i add ${HANGUPCAUSE} in CDR after the Dial command using h extension? I
want to add the Hangup reason of call in userfield of CDR.
http://www.google.com/search?q=asterisk+hangupcause+cdr
Top result... Should do it
Steve
However the two calls are placed, the CDRs and the callerids are set
correctly, we can't hear each other. As I saw in the logs, the problem is
that the calls are placed in the same context, and not being connected (
like one call, but with the variable EXTEN changed ).
I'm really confused
This is obvious for the first Channel ( Channel:
Local/0036701234567@CustomCallOut-1/n ), but how to set on the other party?
I tried with Context: CustomCallOut-2/n but didn't worked.
2011/2/8 Sherwood McGowan sherwood.mcgo...@gmail.com
However the two calls are placed, the CDRs and the
How can I do that, and do it with LCR?
2011/2/8 fai...@vopium.com
Why don't you use single callfile and set CLI and other perameters in
dial-plan as unique as you need?
-Original Message-
From: Tamás Dajka tda...@gmail.com
Sent: Tuesday, February 8, 2011 7:45am
To:
Just verified I faced the same issue once and got it reolved by adding /n like
Channel: [mailto:Local/0036701234567@CustomCallOut-1/n]
Local/0036701234567@CustomCallOut-1/n in you case.
-Original Message-
From: Tamás Dajka tda...@gmail.com
Sent: Tuesday, February 8, 2011 8:49am
To:
Hi All,
First post here. I am dialing out via call file to remote number, when call
is connected a local number is dialed. And on success both calls get bridged
and works fine.
This is a parallel auto dialout application. I want to set a variable as
soon as the local number answers the call, so
hi
i searched a lot but i couldn't find the answer
i have two openvox(fxo/fxs) card so I have 24 ports! on first card i have 12
fxs and on the second i have 12 fxo
i want to then one person calling from dahdi/13 forward it to dahdi/1
when a person calling from dahdi/14 forward it to dahdi/2
when
the M option in your Dial command will execute a macro upon connection,
there's also an option to perform a Gosub...
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
;-)
*keeps his mailing-list police badge in it's box in his office*
(that wasn't directed at you Dan...there was a little
Thanks Faisal. That is it. I was confused by the fact that there is also the
Context, Extension, and Priority in the .call file that should be filled
along with the Channle: local. I found out that the call file first
calls the local channel context and once that is connected then it moves
Hi,
Are sip.conf's defaultexpiry and maxexpiry global?
Or can they be used on a per-extension basis?
I'd like to force some extensions to re-register more frequently than others
(server-side).
Thanks,
Vieri
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Hi,
Suppose you have 2 identical Asterisk servers and 1 alias IP address that you
assign to either one, according to system failures, etc.
Also suppose that all SIP clients register requests go to the alias IP address.
Imagine server1 fails and server2 gets the alias IP address. Correct me if
On 8 Feb 2011, at 14:52, mehran khajavi wrote:
i searched a lot but i couldn't find the answer
.
i have two openvox(fxo/fxs) card so I have 24 ports!
Ok!
on first card i have 12 fxs and on the second i have 12 fxo
i want to then one person calling from dahdi/13 forward it to dahdi/1
when
Tuesday, February 8, 2011, 5:07:29 PM, Vieri wrote:
How can I minimize this time lapse? Can Asterisk notify all SIP
clients in its sip.conf that they need to acknowledge being on-line
or not (thus forcing re-registration in my scenario)?
If you have two identical servers online, it is better
Take a look at High Availability ASTerisk (HAAST) from www.generationd.com
Their software sits between the OS and asterisk, and can failover servers,
switch IP addresses, control external interfaces, etc.
It can run on different hardware (make a cluster from different/cheap boxes),
it allows
Hi,
Thats very simple.
Use sip realtime registration with mysql and heartbit to control switiching.
Regards,
Carlos M Cruz
Em 2011/02/08 16:07, Vieri rentor...@yahoo.com escreveu:
Hi,
Suppose you have 2 identical Asterisk servers and 1 alias IP address that
you assign to either one,
On Tue, Feb 8, 2011 at 8:07 AM, Vieri rentor...@yahoo.com wrote:
Suppose you have 2 identical Asterisk servers and 1 alias IP address that you
assign to either one, according to system failures, etc.
Also suppose that all SIP clients register requests go to the alias IP
address.
This is a
Hi,
Using MeetMee(1234,M) on asterisk 1.6.2.9 (debian/testing) we have
terrible sound: the MOH is unrecognizable and speakers can't be
understood; it sounds ghostly. However the prompts (your are the only
one in this conference, etc.) sound fine.
Our server has a Digium T410P card with two E1
On Mon, 2011-02-07 at 10:44 -0500, Daniel - Asterisk wrote:
Hi Danny,
Could you please let me know what function do I use to get if the
queue is full?
Elder
On Mon, Feb 7, 2011 at 10:42 AM, Danny Nicholas da...@debsinc.com
wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis-David
Mitterrand
Sent: Tuesday, February 08, 2011 10:50 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] terrible MeetMe sound with 1.6.2.9
Hi Users,
I'm planing to implement call completion feature in asterisk 1.8 but having
some issue. I am following this document
https://wiki.asterisk.org/wiki/display/AST/Generic+Call+Completion+Example
I am getting error non-zero error on console. I am using softphone x-lite
This is a problem that is completely stumping me, and my understanding of
Asterisk dialplans tells me this should never be a problem. Moreover, this
scenario works on Asterisk 1.4 but not 1.6.
We have a customer with several Aastra 6731 phones. They want incoming
calls from the PSTN to work and
On Tue, Feb 08, 2011 at 10:59:47AM -0600, Danny Nicholas wrote:
Any idea?
I use mpg123 to play my MOH so I can control the volume (my users complain
that standard MOH is a bit loud).
Forgot to add that our MOH sounds fine when listened to (on the same
extension as MeetMe) with
On Tue, Feb 8, 2011 at 11:02 AM, Ernie Dunbar maill...@lightspeed.cawrote:
Internal calls:
exten = _312,1,Set(CALLERID(name)=Internal call)
exten = _312,n,SIPAddHeader(Alert-Info: info=Bellcore-dr2)
exten = _312,n,Dial(SIP/username2,20)
exten = _312,n,Voicemail(312,u)
exten =
On Tue, Feb 8, 2011 at 11:04 AM, Louis-David Mitterrand
vindex+lists-asterisk-us...@apartia.org wrote:
Forgot to add that our MOH sounds fine when listened to (on the same
extension as MeetMe) with MusicOnHold(default). So it's not a MOH
problem as speakers in the MeetMe conference are
On Tue, Feb 08, 2011 at 11:09:19AM -0600, Warren Selby wrote:
On Tue, Feb 8, 2011 at 11:04 AM, Louis-David Mitterrand
vindex+lists-asterisk-us...@apartia.org wrote:
Forgot to add that our MOH sounds fine when listened to (on the same
extension as MeetMe) with MusicOnHold(default). So it's
On Tuesday 08 February 2011 06:34:56 Sherwood McGowan wrote:
On Tue, Feb 8, 2011 at 6:01 AM, fai...@vopium.com wrote:
But if you are getting calls all the way on VoIP then you can have
calls in HD audio using HD audio codec on all locations (Server and
Client). In that case you either need
Thanks, I will check our that. It seems M macro would work.
-dani
On Tue, Feb 8, 2011 at 7:02 AM, Sherwood McGowan sherwood.mcgo...@gmail.com
wrote:
the M option in your Dial command will execute a macro upon connection,
there's also an option to perform a Gosub...
Hello all,
Just hoping to get some opinions from folks that have actually used the
Rhino R4FXO-EC. Looking for user experiences, good or bad. This looks
like a nice unit and I have a need for exactly this config, 4FXO and EC
TIA,
JohnM
--
Hi,
For future reference, it might be useful to notice (from SIP 3.1 Admin
Manual):
attendant/ attributes are only available to SoundPoint 320/330, 430, 550,
560, 600, 601, 650 and 670 phones only.
For a 3.1.3-enabled 501, has someone been able monitor a third status beyond
Idle, OnCall ones ? I
On Thursday, February 10, 2011 at 8:00AM CST (GMT-5), two servers that
provide community services will be upgraded with new software releases:
* wiki.asterisk.org will be upgraded to Confluence 3.4.8. This upgrade
should take less than 20 minutes.
* code.asterisk.org will be upgraded to
Hello Everyone!
I've hit a bit of a roadblock and I am hoping that someone might point
me in the right direction.
I am using Asterisk 1.2.4 - I do not have the option of updating it,
please do not waste your time telling me to =)
I am using PERL AGI scripts to maintain an active calls count
Hello all.
I have a asterisk implementation (asterisknow 1.7.1), with a card with 2 FXO
interfaces.
When I call (or receive a call) from the pstn, I ear echo. This happens if I
use a softphone or IP phone, and does not happens if the call is internal.
Can you help me with this issue?
Best
Hello All,
I was wondering if anyone's tried to use OR currently use the Microsoft
Speech Server or their UCMA 3.x SDK etc. as their ASR/TTS backend/engines
etc. If yes, then what's their experience? Please Note, this does NOT need
to be integrated with Asterisk ala MRCP or some module/plugin
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