Displaying 20 results from an estimated 6000 matches similar to: "AEL2: If-then-else not permitted in Switch-Case"
2009 Apr 07
1
AEL2, BASE64_DECODE and hexadecimal
Hi,
I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in an
AEL2 file like this :
SendText(${BASE64_DECODE(DQ==)});
Value sent (8 bytes long) is very strange :
Content-Type: text/plain;charset=UTF-8
Content-Length: 8
?ez?==
Any workaround ?
Regards
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2009 Feb 05
6
Newbie query: how to write priority n+101
Hi All,
Asterisk 1.4.12 on CentOS 5
Sorry for a question that I'm guessing is obvious to most of you.
I'm trying to revamp my dialplan. When I first created it, I had
something like:
exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => s,2,Dial(${rgMain},${RINGTIME},t)
exten => s,3,VoiceMail(main at default)
exten => s,103,VoiceMail(main at default)
2009 Jul 13
1
#exec in #include'd file
Hi,
Is Asterisk supposed to evaluate #exec's in an #include'd file?
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de
--
2009 May 11
1
Support of /* */ comments in ael.vim
Hello,
It seems /* */ comments are not supported in ael.vim (which brings AEL
syntax-highlighting to vim).
Is it hard to add this feature and have uploaded in vim extensions
downloading site ?
Regards
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2009 May 24
7
Asterisk, SQL Database Update
Is there any method in Asterisk to enable the updating process
into another SQL database through entering IVR options during the call.
Thanks a lot.
_________________________________________________________________
Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!
2009 Jan 08
4
AEL question: testing channel variables
Hi!
I use the following condition:
if (${FOOBAR}=YES) {
...
}
The problem is, that if FOOBAR is not defined at all Asterisk generates
a warning:
WARNING[11982]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
=YES
Of course I could use the following code, but this bloats up the code:
if (${EXISTS(${FOOBAR})}) {
2009 Jun 12
2
Current possible values for DIALSTATUS?
Hi,
As of v 1.6.1.1, can anyone tell me what the current possible values for
DIALSTATUS could be? I found
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe
it is outdated since there is no FAIL or FAILED in this list.
Thanks!
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2009 Apr 16
7
How to send "404 Not found" SIP reply?
Hi,
I am trying to send "404 Not found" reply, without any luck with the
following:
exten => 555,1,Playback(you-dialed-wrong-number,noanswer)
exten => 555,n,Playback(check-number-dial-again,noanswer)
exten => 555,n,Congestion()
However the above results in "500 Service Unavailable" being send out.
What would be the correct application/function to generate "404
2009 Jan 10
1
Local channel Help required
Hi All,
I am using asterisk 1.4 branch on server.
Here is a my dialplan.
i have set the incoming route to incoming context, and then i have set dial
with local channel,
The call comes to my server and the call is routed to matched case, so my
phone 1001 ring for 30 seconds.
If i got the NOANSWER then the channel is not passing to next priority.
I need to pass that channel to the next priority of
2009 Jul 22
3
ExecIf and empty variables (early evaluation)
Imagine that you have this code:
exten => _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
If ${QueueName} happens to be unset, this will cause a warning:
[Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187
queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an
argument: queuename
The obvious solution:
exten => _X!,n,ExecIf($["${QueueName}" !=
2009 Jul 14
2
How to block inbound call with Asterisk?
Guys,
How would you block inbound call's? for example person who is calling me is
212-555-1212, and I would like to do not receive the calls from this person
and give them busy tone.
What should I write in asterisk config files? and in to witch file should I
write it???
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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 Jul 24
3
Goto from a feature macro is not working?
Hello,
I'm trying to implement multi-party calls according to these
instructions:
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
They are almost working, except that the Goto at the end of
[dynamic-nway-start] doesn't seem to work. When I turn verbosity up a
bit, I get something like this in my error log:
== Channel 'SIP/SWG-0085a180' jumping out of macro
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 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 Jun 02
2
SIP Response 181 - Is it possible in Asterisk?
Hello all,
I have being trying to replicate the following call scenario with my
Asterisk box: http://www.tech-invite.com/Ti-sip-service-8.html
<http://www.tech-invite.com/Ti-sip-service-8.html>
I have a situation that I have to notify the calling party that the call is
being forwarded to another number. So far, in the tests that I made, calling
from a SIP extension to another SIP
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