Displaying 20 results from an estimated 6000 matches similar to: "Determining extension's sip.conf default mailbox"
2009 Jun 04
2
broken pipe in perl agi
Hi gang,
Since I'm getting no joy from device_Status or SIPPEER in
1.4.26-rc1, I thought I would do an AGI to read my hints and check for line
in use that way. The AGI works fine from a prompt, but returns the dreaded
"utils.c:966 ast_carefulwrite: write() returned error: Broken pipe" when I
try to run it from the dialplan. Here is my dialplan snippet;
2009 Jul 23
5
Music on hold based on user
Hi
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play music
1, if another from Technical Support put on hold the call play music 3,
something like this
Admin - Music1
Contrallors - Music 2
Technical Support - Music 3
Thanks
--
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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 Mar 16
2
Busy on SIP
Hi,
I have a question. How can I configure my sip.conf to make a SIP phone busy
on incoming and outcoming calls? I explain my problem.
When SIP phone receive a call and then I try to call that phone, I find it
busy.
When SIP phone make a call and I try to call that phone, I find it avaible
and it rings but I want to find it busy.
I configure sip.conf like following:
[10]
type=friend
qualify=yes
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 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
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 Dec 04
1
Get Queue values from dialplan (Was: queue_variables() function)
2009/12/4 Olivier <oza-4h07 at myamail.com>
> Hello,
>
> Has someone successfully used this QUEUE_VARIABLES() function (in
> 1.6.2-rc7) ?
> I tried to use it as I'm using SIPPEER() but without success.
>
> A previous question about it remainded unanswered (
> http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466).
>
> Regards
>
How can