Displaying 20 results from an estimated 6000 matches similar to: "Aastra phone crashes with Asterisk 1.6"
2009 Jun 11
2
OT - Aastra phones provisioning
Hi,
I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new
Aastra SIP phones can be auto-provisioned when config files are stored in a
specific TFTP subdirectory instead of TFTP root directory.
For instance, TFTP root directory is /srv/tftp.
When config files are stored in /srv/tftp, a new Aastra can find its config
files.
When config files are stored in /srv/tftp/aastra,
2009 Jan 18
3
Using a sidecar? Ideas?
I'm looking for some ideas of people who have setup a sidecar (eg: Aastra
560M).
Obviously it's handy for BLF (to see who's on a call)...but what else?
Anyone want to share interesting things they've done with a sidecar?
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2006 Feb 16
3
Firmware version 1.3.1 released for Aastra IPphones
There is no release note, just a text file that says
AASTRA TELECOM INC.
February 2006
FC-000046-01-07.st - 9133i Generic SIP Firmware 1.3.1.1095 for customer
release.
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gareth
Owen
Sent: 15 February 2006 02:00
To: asterisk-users@lists.digium.com
2006 Mar 17
4
Aastra Questions
Hi,
Does anyone have experience with Asterisk and the aastra 9112i or
9133i phones? I am looking at purchasing some, and was curious how
quality, and stability was with them.
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 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 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 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 Jan 21
1
SIP realtime status...
Since 1.4.22 realtime status for sip peers seems to be broken. If I do
a "sip show peers" from the CLI I get this:
2001/2001 192.168.2.234 D 5060 UNKNOWN
Cached RT
It is arbitrary which peers will say OK and which will say UNKNOWN and
it changes over time. This is a problem with an application like the
Asternic Flash panel because it uses the peer
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 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 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 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
2008 Sep 14
1
MoH with an Aastra 9112i
Hello,
I have some Aastra 9112i's in production that almost function
flawlessly. The problem I'm having is when a caller is put on hold they
do not hear hold music. If they are on hold for too long (~ a minute?)
they are hung up on.
All other phones including Aastra 480i and Sipura/Linksys ATAs all seem
to be working fine.
Is this a quirk anyone else has experienced? Any