Displaying 20 results from an estimated 7000 matches similar to: "No subject"
2013 Mar 15
0
No subject
;
; Display certain channel variables every time a channel-oriented
; event is emitted:
;
;channelvars =3D var1,var2,var3
So if you want fu_callerid, set:
channelvars =3D fu_callerid
And, once that variable is set, you should get a NewExten event, you
should see the following key/value pair:
ChanVariable(SIP/1234-00000001): fu_callerid=3Dfoobar
--=20
Matthew Jordan
Digium, Inc. | Engineering
2013 May 07
1
Get Channel Variables in AMI Event NewExten
Hi, I'm stucked in situation, and look for a work around if possible in Asterisk.
I have a dialplan,
[default]
exten => 111222,n,Set(fu_callerid=141688xyxzz)
exten => _X.,n,NoOp(Callerid ${fu_callerid})
exten => _X.,n,wait(2)
exten => _X.,n,Answer()
?
When, ?Answer Application is called AMI Event is triggered like this..
? ? ? ? ? 'Event' => 'Newexten',
? ? ? ?
2009 Jul 20
0
No subject
faced this exact same problem a few times on more than one servers and it
was 1) dialplan issue which was not hanging up the zap channels correctly 2)
using more than 8 spans on a server. Asterisk can't handle more than 96 zap
channels on T1s. FXO/FXS combinations can vary the number of spans but if
you know what I mean by spans, in production don't use more than 6 spans.
On 2010-03-17
2013 Mar 15
0
No subject
kernel-headers-2.6.32-279.el6.x86_64.rpm<br>
<br>
Alec Davis<br>
<br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href=3D"http://www.api-digital.c=
om" target=3D"_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us
2011 Apr 12
0
No subject
supported, beside Idle, On call and Ringing ?
Can we expect this list to match DEVICE_STATE's one (UNKNOWN | NOT_INUSE |
INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD)
> Might be worth seeing if other phones do the same.
>
> S
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by
2009 Jul 20
0
No subject
at least once a week I receive such an attack coming from a different ip.
I will read the articles. Thanks again to everyone.
Regards,
Rodrigo Lang.
2010/6/29 Kenny Watson <kwatson at geniusgroupltd.com>
> Hi, you can use fail2ban
>
2009 Jul 20
0
No subject
device somewhere in your communication path, and since voice is picked up as
DTMF, some device is also set to listen for inband DTMF.
What is the origination source of incoming calls to your system?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-08 4:24 PM, "das sandesh" <sandesh440 at gmail.com> wrote:
Hi,
We have few systems with asterisk 1.4.22.1 and we use sip
2009 Jul 20
0
No subject
mailboxes).
Are you certain that removing either 612 or 610 mailbox would keep Asterisk
from complaining ?
>
> However, the MWI does not indicate voice mails for 610 and I keep seeing
> this error message:
>
> ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox
> 610 in context a10
>
> However, mailbox 610 is clearly defined in voicemail.conf:
>
2009 Jul 20
0
No subject
-uzzi
PS: If you're not seeing any connection information, be sure to double-check
the IP address is correct. Learned that lesson the hard way =\
On Sun, Jan 31, 2010 at 5:51 PM, Jim Rosenberg <jr at amanue.com> wrote:
> Let's say I have two Asterisk boxes, A and B. I am trying to get A to do
> SIP registration on B, so an extension for A can dial SIP phones covered by
>
2009 Jul 20
0
No subject
/var/lib/asterisk/sounds/soundfile.alaw
/var/lib/asterisk/sounds/soundfile.wav
to go from alaw to mp3, first convert to wav, then use lame <options>
/var/lib/asterisk/sounds/soundfile.wav
/var/lib/asterisk/sounds/soundfile.mp3
sox looks like it can ogg/vorbis, but mine doesn't list mp3. You might fetch
the source for sox and see if it can do mp3; lame is probably
just as easy to obtain
2009 Jul 20
0
No subject
have adaptors compatible with Asterisk, but explicitly say in the product
titles that they're unlocked, which I think is the key.
On Thu, Dec 17, 2009 at 4:16 AM, Brian Cline <Brian at nw.brian.fm> wrote:
> Hello,
>
> I'm running Asterisk v1.6.1.11 internally with a few Linksys SIP
> phones and will be receiving a machine containing a Dialogic card
> for a
2017 Oct 04
0
Samba upgrade from 4.6 to 4.7
On 10/4/2017 9:27 AM, Harsh Kukreja wrote:
> Thanks for the response. Does that mean that I will have to rebuild
> the samba again after upgrading Ubuntu 12.04 to 14.04
>
> *Harsh Kukreja *Systems Administrator
>
> **International University of Namibia**Tel: 061-4336000 - E-mail:
> h.kukreja at ium.edu.na <mailto:h.kukreja at ium.edu.na> - Web:
>
2004 Feb 19
1
Process R segmentation with strsplit() (PR#6601)
Getting a crash with R1.8.1 on windows 2000 an linux with the strsplit.
Version:
platform = i386-pc-mingw32
arch = i386
os = mingw32
system = i386, mingw32
status =
major = 1
minor = 8.1
year = 2003
month = 11
day = 21
language = R
Version:
platform = i386-pc-mingw32
arch = i386
os = mingw32
system = i386, mingw32
status =
major = 1
minor = 8.1
year = 2003
month = 11
day =
2005 Jan 21
0
Help DIALSTATUS gives ANSWER when line is BUSY?
I'm running Asterisk CVS-v1-0-12/20/04.
I'm using PHP with Manager API Here is the code:
####################################################################
# Make call
####################################################################
$socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout);
if (!$socket) {
echo "$errstr ($errno)<br /\n";
} else {
2007 Dec 10
0
diferents events between ast1.2 & ast1.4 ??
Hi all,
I'm new in the list, and I have a problem upgrading from asterisk 1.2 to
asterisk 1.4:
There is a diference from asterisk1.2 to asterisk1.4 in AMI events.
When I do a call to a queue (with the same extensions.conf dial plan)
with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4
apper only 2.
It is normal? anyone knows it? what is the reason?
I
2018 Jan 16
0
User Permissions issue
Hi Harsh,
>
> Thanks for your advise I will not use these wordings here.
thanks!
> Please check the result below when I run the command on the DC-1 when
> DC-2 is off or on
> smbclient -k //IUMSVRAPP01/Pastel12 -d 9
> ...
> session setup failed: NT_STATUS_INVALID_PARAMETER_MIX
Looking at this message, I would start with doing some cleanup in your
smb.conf. I would trim
2018 Jan 17
0
User Permissions issue
Hi Harsh,
> Thanks for the suggestion to trim the smb.conf after which the DC-1 is
> connecting to the Windows Server 2008 shared folder smbclient -k
> //IUMSVRAPP01/Pastel12 -d 9
> and DC-2 is also connecting after using the DNS name of the Windows server.
>
> *You'd better switch your DNS to Bind-DLZ. Internal DNS is not that good
> for larger site (looking at your DNS
2018 Jan 19
1
User Permissions issue
Hi Denis
I have upgraded my samba DC-1 from 4.6.12 to 4.7.4 which has solved the
replication issues between DC-1 and DC-2. Now both the DC's are running on
4.7.4.
Like Rowland said previously, you should remove all RODC that have been
installed prior to Samba 4.7. There are many fixes that have been added
since 4.6.
Before I remove my RODC's I like to clear out few doubts:
1. Instead of
2017 Oct 04
3
Samba upgrade from 4.6 to 4.7
Hello
I have a Samba 4.6.7 server installed with the sernet package on Ubuntu
12.04 precise which I like to upgrade to 4.7 and also I want to do a
release upgrade for Ubuntu from 12.04 to 14.04.
Before I continue I like know if I should first do a release upgrade for
Ubuntu or do I have to upgrade Samba to 4.7 first and then run the release
upgrade.
Thanks n Regards
*Harsh Kukreja *Systems
2017 Oct 28
1
ADC 4.7.0 KCC replication failing with PDC 4.6.8
Hi Rowland
Thanks for the link https://wiki.samba.org/index.php/Samba_AD_DC_Port_Usage
It should fix the Firewall authentication problem.
I like to find out if I can rename the Domain name of the DC when I move it
to the new DC like currently it is IUMNET.EDU.NA <http://iumnet.edu.na/> but
my GApps domain name is IUM.EDU.NA <http://ium.edu.na/> and I like to use
the same on my new