Displaying 20 results from an estimated 2000 matches similar to: "SIP Response 181 - Is it possible in Asterisk?"
2009 Jun 03
0
RES: RES: SIP Response 181 - Is it possible in A steri sk?
Hello Philipp and All,
My scenario is a bit different than the one I had explained before. I'm
sorry.
Let's suppose I have someone calling one of my Asterisk clients. This
asterisk client has CFB (Call Forward Busy) activated. The forward number is
a Voice Mail System, however is not a Asterisk's Voice Mail.
It is a third party Voice Mail System, that has a SIP Trunk with my
2009 Jan 27
2
Module res_odbc is not loading
Hi,
I have remove the comment defor res_odbc.so and res_config_odbc.so in my
modules.conf, but the module is still not loading
when I do:
module show like odbc
I have o module returned
anybody knows why?
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2009 Jan 07
5
recommendation for German sound files
Hi!
http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German
lists a plenty of sound files for German.
Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon).
thanks
klaus
2009 Jan 26
2
custom cdr userfiled
Dear,
I added new field to cdr table , named "service" and type varchar(20),
but in extensions.conf with the following command, nothing to be saved.
exten => _X.,1,Set(CDR(service)=OUT)
does asterisk support this ability ?
is any setting must be changed, before that ?
best
Mani
2009 Nov 10
2
Hangup
Hi, is it possible to hangup a channel from another channel?
I want to finish a call from another channel, but if I put
exten => h,n,HangUp(channelname)
it doesn't hangup... Is that correct?
Thanks,
_________________________________________________________________
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2009 May 24
7
Asterisk, SQL Database Update
Is there any method in Asterisk to enable the updating process
into another SQL database through entering IVR options during the call.
Thanks a lot.
_________________________________________________________________
Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!
2009 Feb 09
2
InUse&Ringing
Hello,
I'm just wondering if anyone has fixed the 'InUse&Ringing' problem.
* v1.4.23.1
Ta
2009 Mar 04
5
AEL2: If-then-else not permitted in Switch-Case
I just want to confirm but it seems that if-then-else is not permitted
in case structure.
It was not really documented but it seems to be the case.
Can anyone confirm?
switch(${DIALSTATUS})
{
case NOANSWER:
{
// if-then-else not permitted
If (${ael-var} = 1)
{
Playback(beep);
2009 May 17
4
Can YOU find a trailing parenthesis?
On 1.6.1, I must be losing my eyesight:
[internal]
include => outbound-pstn
.............
include => meetme ; 2663
include => setup-meetme-conf-room ; 6000xxxYYYY
[setup-meetme-conf-room]
exten => _6000XXXNXXX,n,Set(Time-in-secs="${STRFTIME(${EPOCH},,%s}" )
........
CLI:
-- Starting simple switch on 'DAHDI/1-1'
[2009-05-17 14:54:49]
2009 Jan 13
9
FWD and Asterisk
I have an account with FWD and I have configured my SIP.conf with
[fwd]
type=friend
secret=password
username=901835
host=fwd.pulver.com
But when I am trying to dial out my own DID , I dont see any call landing in
asterisk.
In extension.conf (vicidial) file I have
exten => 2062036895 ,1,Ringing()
exten => 2062036895 ,2,Wait(1)
exten => 2062036895 ,3,Answer()
exten => 2062036895
2009 Feb 02
2
Invalid Extension
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
CLI Output :
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
vicidialnow*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from
2009 Feb 11
3
call forward all except the extension it is forwarded to
I would like to know if I can set Call Forwarding on an extension but allow direct calls from the extension it is forwarded to.
Example:
Extension 100 sets call forwarding (all) to extension 101.
All calls to 100 are immediately forwarded to 101 as expected.
However, if 101 tries to transfer a call to 100 or tries to call 100 directly, it sounds "busy" because it obviously goes into
2009 Apr 16
1
Remote BLF / hint on IAX2 trunk
Hi all,
I have a question: how can I see hints of a remote Asterisk in IAX2 trunk??
I want to set BLF on my phones to look state of other phones also in other
Asterisk server.
Someone have any idea or solution?
I use Asterisk 1.4.24.
Thanks all
Marco
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2009 May 22
1
/etc/asterisk/startup.d
Does anybody think it would make sense for /etc/init.d/asterisk
to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk
did?
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP
2009 Jan 07
2
\iaxclient-2.0.2 compile problem
Hi,
I had downlaoded iaxclient-2.0.2 and complie project
*\iaxclient-2.0.2\contrib\win\vs2005*
**
It gives many83 fatal and file missing error of file missing
Error 1 fatal error C1083: Cannot open include file: 'portaudio.h': No such
file or
directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\portmixer\px_win_wmme\px_win_wmme.c
40
Error 2 fatal error C1083: Cannot open
2009 Jun 12
1
AmooCon video recordings online
JFYI and slightly off-topic:
All of the videos we recorded at AMOOCON open-source VoIP conference
(Rostock, Germany, May 4-5) are now available on the web site:
http://www.amoocon.com/
All of them are available in different qualities and formats,
including Quicktime 7, versions for the iPhone and iPod and h.264
which IIRC can be played in MPlayer etc.
100 GB in total. :-)
Philipp Kempgen
2009 Jul 24
4
Web Browser Pop-up
Heelo,
I currently search a program that can make a web browser Pop-up on an
incoming call on a specific URL like :
http://directorie.ch?CALLNUMBER:00451849799
I have found ADM, but it's a bit more complex for my purpose an it's not
very stable.
Do you know a simple software for that ?
An other part of my project is to eneable click-to-call from a web page, do
you know a kind of
2009 May 19
8
Ghost ??
We are using asterisk and sometime when our guys are on call , they hear
some voice of person and amazingly that person is NOT from our center.
Any one faced this kind of thing ?
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2009 Jun 09
5
voicemail
Has anyone set it up so that an inside call and an outside call get
different unavailable messages?
j
2009 Feb 05
6
Newbie query: how to write priority n+101
Hi All,
Asterisk 1.4.12 on CentOS 5
Sorry for a question that I'm guessing is obvious to most of you.
I'm trying to revamp my dialplan. When I first created it, I had
something like:
exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => s,2,Dial(${rgMain},${RINGTIME},t)
exten => s,3,VoiceMail(main at default)
exten => s,103,VoiceMail(main at default)